ari-stubs: Fix more local anchor references

Also allow CreateDocs job to be run manually with default branches.

.github/ISSUE_TEMPLATE/bug-report.yml

@@ -0,0 +1,87 @@
+name: Bug
+description: File a bug report
+title: "[bug]: "
+labels: ["bug", "triage"]
+#assignees:
+#  - octocat
+body:
+  - type: markdown
+    attributes:
+      value: |
+        Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.
+
+        A good first step is for you to review the Asterisk Issue Guidelines if you haven't already. The guidelines detail what is expected from an Asterisk issue report.
+
+        Then, if you are submitting a patch, please review the Patch Contribution Process.
+
+        Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.
+
+        Please note that by submitting data, code, or documentation to Sangoma through GitHub, you accept the Terms of Use present at
+        https://www.asterisk.org/terms-of-use/.      
+        Thanks for taking the time to fill out this bug report!
+  - type: dropdown
+    id: severity
+    attributes:
+      label: Severity
+      options:
+        - Trivial
+        - Minor
+        - Major
+        - Critical
+        - Blocker
+    validations:
+      required: true
+  - type: input
+    id: versions
+    attributes:
+      label: Versions
+      description: Enter one or more versions separated by commas.
+    validations:
+      required: true
+  - type: input
+    id: components
+    attributes:
+      label: Components/Modules
+      description: Enter one or more components or modules separated by commas.
+    validations:
+      required: true
+  - type: textarea
+    id: environment
+    attributes:
+      label: Operating Environment
+      description: OS,  Disribution, Version, etc.
+    validations:
+      required: true
+  - type: dropdown
+    id: frequency
+    attributes:
+      label: Frequency of Occurrence
+      options:
+        - "Never"
+        - "One Time"
+        - "Occasional"
+        - "Frequent"
+        - "Constant"
+  - type: textarea
+    id: description
+    attributes:
+      label: Issue Description
+    validations:
+      required: true
+  - type: textarea
+    id: logs
+    attributes:
+      label: Relevant log output
+      description: Please copy and paste any relevant log output. This will be automatically formatted into code, so no need for backticks.
+      render: shell
+  - type: markdown
+    attributes:
+      value: |
+        [Asterisk Issue Guidelines](https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines)
+  - type: checkboxes
+    id: guidelines
+    attributes:
+      label: Asterisk Issue Guidelines
+      options:
+        - label: Yes, I have read the Asterisk Issue Guidelines
+          required: true

.github/ISSUE_TEMPLATE/config.yml

@@ -0,0 +1,8 @@
+blank_issues_enabled: false
+contact_links:
+  - name: Asterisk Community Support
+    url: https://community.asterisk.org
+    about: Please ask and answer questions here.
+  - name: Feature Requests
+    url: https://github.com/asterisk/asterisk-feature-requests/issues
+    about: Please submit feature requests here.

.github/ISSUE_TEMPLATE/improvement.yml

@@ -0,0 +1,27 @@
+name: Improvement
+description: Submit an improvement to existing functionality
+title: "[improvement]: "
+labels: ["improvement", "triage"]
+body:
+  - type: markdown
+    attributes:
+      value: |
+        Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.
+
+        A good first step is for you to review the Asterisk Issue Guidelines if you haven't already. The guidelines detail what is expected from an Asterisk issue report.
+
+        Then, if you are submitting a patch, please review the Patch Contribution Process.
+
+        Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.
+
+        Please note that by submitting data, code, or documentation to Sangoma through GitHub, you accept the Terms of Use present at
+        https://www.asterisk.org/terms-of-use/.      
+        Thanks for taking the time to fill out this bug report!
+  - type: textarea
+    id: description
+    attributes:
+      label: Improvement Description
+      description: Describe the improvement in as much detail as possible
+    validations:
+      required: true
+      

.github/ISSUE_TEMPLATE/new-feature.yml

@@ -0,0 +1,27 @@
+name: New Feature Submission
+description: Submit a New Feature
+title: "[new-feature]: "
+labels: ["new-feature", "triage"]
+body:
+  - type: markdown
+    attributes:
+      value: |
+        Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.
+
+        A good first step is for you to review the Asterisk Issue Guidelines if you haven't already. The guidelines detail what is expected from an Asterisk issue report.
+
+        Then, if you are submitting a patch, please review the Patch Contribution Process.
+
+        Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.
+
+        Please note that by submitting data, code, or documentation to Sangoma through GitHub, you accept the Terms of Use present at
+        https://www.asterisk.org/terms-of-use/.      
+        Thanks for taking the time to fill out this bug report!
+  - type: textarea
+    id: description
+    attributes:
+      label: Feature Description
+      description: Describe the new feature in as much detail as possible
+    validations:
+      required: true
+

.github/workflows/CherryPickTest.yml

@@ -0,0 +1,167 @@
+name: CherryPickTest
+run-name: "Cherry-Pick Tests for PR ${{github.event.number}}"
+on:
+  pull_request_target:
+    types: [ labeled ]
+
+concurrency:
+  group: ${{github.workflow}}-${{github.event.number}}
+  cancel-in-progress: true
+
+env:
+  PR_NUMBER:          ${{ github.event.number }}
+  MODULES_BLACKLIST:  ${{ vars.GATETEST_MODULES_BLACKLIST }} ${{ vars.UNITTEST_MODULES_BLACKLIST }}
+
+jobs:
+  IdentifyBranches:
+    name: IdentifyBranches
+    if: ${{ github.event.label.name == vars.CHERRY_PICK_TEST_LABEL }}
+    outputs:
+      branches:     ${{ steps.getbranches.outputs.branches }}
+      branch_count: ${{ steps.getbranches.outputs.branch_count }}
+    runs-on: ubuntu-latest
+    steps:
+      - name: Remove Trigger Label, Add InProgress Label
+        env:
+          GH_TOKEN:          ${{ secrets.GITHUB_TOKEN }}
+        run: | 
+          gh pr edit --repo ${{github.repository}} \
+            --remove-label ${{vars.CHERRY_PICK_TEST_LABEL}} \
+            --remove-label ${{vars.CHERRY_PICK_CHECKS_PASSED_LABEL}} \
+            --remove-label ${{vars.CHERRY_PICK_CHECKS_FAILED_LABEL}} \
+            --remove-label ${{vars.CHERRY_PICK_GATES_PASSED_LABEL}} \
+            --remove-label ${{vars.CHERRY_PICK_GATES_FAILED_LABEL}} \
+            --remove-label ${{vars.CHERRY_PICK_TESTING_IN_PROGRESS}} \
+            ${{env.PR_NUMBER}} || :
+
+      - name: Get cherry-pick branches
+        uses: asterisk/asterisk-ci-actions/GetCherryPickBranchesFromPR@main
+        id: getbranches
+        with:
+          repo:                    ${{github.repository}}
+          pr_number:               ${{env.PR_NUMBER}}
+          cherry_pick_regex:       ${{vars.CHERRY_PICK_REGEX}}
+          github_token:            ${{secrets.GITHUB_TOKEN}}
+
+      - name: Check Branch Count
+        if: ${{ steps.getbranches.outputs.branch_count > 0 }}
+        env:
+          GH_TOKEN:          ${{ secrets.GITHUB_TOKEN }}
+        run: |
+          gh pr edit --repo ${{github.repository}} \
+          --add-label ${{vars.CHERRY_PICK_TESTING_IN_PROGRESS}} \
+          ${{env.PR_NUMBER}} || :
+
+  CherryPickUnitTestMatrix:
+    needs: [ IdentifyBranches ]
+    if: ${{ needs.IdentifyBranches.outputs.branch_count > 0 && ( success() || failure() ) }}
+    continue-on-error: false
+    strategy:
+      fail-fast: false
+      matrix:
+        branch: ${{ fromJSON(needs.IdentifyBranches.outputs.branches) }}
+    runs-on: ubuntu-latest
+    steps:
+      - name: Run Unit Tests for branch ${{matrix.branch}}
+        uses: asterisk/asterisk-ci-actions/AsteriskUnitComposite@main
+        with:
+          asterisk_repo:     ${{github.repository}}
+          pr_number:         ${{env.PR_NUMBER}}
+          base_branch:       ${{matrix.branch}}
+          is_cherry_pick:    true
+          modules_blacklist: ${{env.MODULES_BLACKLIST}}
+          github_token:      ${{secrets.GITHUB_TOKEN}}
+          unittest_command:  ${{vars.UNITTEST_COMMAND}}
+
+  CherryPickUnitTests:
+    needs: [ IdentifyBranches, CherryPickUnitTestMatrix ]
+    if: ${{ needs.IdentifyBranches.outputs.branch_count > 0 && ( success() || failure() ) }}
+    runs-on: ubuntu-latest
+    steps:
+      - name: Check unit test matrix status
+        env:
+          RESULT:    ${{needs.CherryPickUnitTestMatrix.result}}
+          GH_TOKEN:  ${{ secrets.GITHUB_TOKEN }}
+        run: |
+          case $RESULT in
+            success)
+              gh pr edit --repo ${{github.repository}} \
+                --add-label ${{vars.CHERRY_PICK_CHECKS_PASSED_LABEL}} \
+                ${{env.PR_NUMBER}} || :
+              echo "::notice::All tests passed"
+              exit 0
+              ;;
+            skipped)
+              gh pr edit --repo ${{github.repository}} \
+                --remove-label ${{vars.CHERRY_PICK_TESTING_IN_PROGRESS}} \
+                --add-label ${{vars.CHERRY_PICK_CHECKS_FAILED_LABEL}} \
+                ${{env.PR_NUMBER}} || :
+              echo "::notice::Unit tests were skipped because of an earlier failure"
+              exit 1
+              ;;
+            *)
+              gh pr edit --repo ${{github.repository}} \
+                --remove-label ${{vars.CHERRY_PICK_TESTING_IN_PROGRESS}} \
+                --add-label ${{vars.CHERRY_PICK_CHECKS_FAILED_LABEL}} \
+                ${{env.PR_NUMBER}} || :
+              echo "::error::One or more tests failed ($RESULT)"
+              exit 1
+          esac
+
+  CherryPickGateTestMatrix:
+    needs: [ IdentifyBranches, CherryPickUnitTests ]
+    if: ${{ success() }}
+    continue-on-error: false
+    strategy:
+      fail-fast: false
+      matrix:
+        branch: ${{ fromJSON(needs.IdentifyBranches.outputs.branches) }}
+        group: ${{ fromJSON(vars.GATETEST_LIST) }}
+    runs-on: ubuntu-latest
+    steps:
+      - name: Run Gate Tests for ${{ matrix.group }}-${{matrix.branch}}
+        uses: asterisk/asterisk-ci-actions/AsteriskGateComposite@main
+        with:
+          test_type:         Gate
+          asterisk_repo:     ${{github.repository}}
+          pr_number:         ${{env.PR_NUMBER}}
+          base_branch:       ${{matrix.branch}}
+          is_cherry_pick:    true
+          modules_blacklist: ${{env.MODULES_BLACKLIST}}
+          github_token:      ${{secrets.GITHUB_TOKEN}}
+          testsuite_repo:    ${{vars.TESTSUITE_REPO}}
+          gatetest_group:    ${{matrix.group}}
+          gatetest_commands: ${{vars.GATETEST_COMMANDS}}
+
+  CherryPickGateTests:
+    needs: [ IdentifyBranches, CherryPickGateTestMatrix ]
+    if: ${{ success() || failure() }}
+    runs-on: ubuntu-latest
+    steps:
+      - name: Check test matrix status
+        env:
+          RESULT:    ${{needs.CherryPickGateTestMatrix.result}}
+          GH_TOKEN:  ${{ secrets.GITHUB_TOKEN }}
+        run: |
+          gh pr edit --repo ${{github.repository}} \
+            --remove-label ${{vars.CHERRY_PICK_TESTING_IN_PROGRESS}} \
+            ${{env.PR_NUMBER}} || :
+          case $RESULT in
+            success)
+              gh pr edit --repo ${{github.repository}} \
+                --add-label ${{vars.CHERRY_PICK_GATES_PASSED_LABEL}} \
+                ${{env.PR_NUMBER}} || :
+              echo "::notice::All Testsuite tests passed"
+              exit 0
+              ;;
+            skipped)
+              echo "::error::Testsuite tests were skipped because of an earlier failure"
+              exit 1
+              ;;
+            *)
+              gh pr edit --repo ${{github.repository}} \
+                --add-label ${{vars.CHERRY_PICK_GATES_FAILED_LABEL}} \
+                ${{env.PR_NUMBER}} || :
+              echo "::error::One or more Testsuite tests failed ($RESULT)"
+              exit 1
+          esac

.github/workflows/CreateDocs.yml

@@ -0,0 +1,123 @@
+name: CreateDocs
+on:
+  workflow_dispatch:
+    inputs:
+      branches:
+        description: "JSON array of branches: ['18','20'] (no spaces)"
+        required:    false
+        type:        string
+  schedule:
+#           Times are UTC
+    - cron: '0 04 * * *'
+
+env:
+  ASTERISK_REPO:     ${{ github.repository }}
+  GITHUB_TOKEN:      ${{ secrets.GITHUB_TOKEN }}
+  GH_TOKEN:          ${{ secrets.GITHUB_TOKEN }}
+  DEFAULT_BRANCHES:  ${{ vars.WIKIDOC_BRANCHES }}
+  INPUT_BRANCHES:    ${{ inputs.branches }}
+
+jobs:
+  
+  CreateDocsDebug:
+    runs-on: ubuntu-latest
+    outputs:
+      manual_branches:     ${{ steps.setup.outputs.manual_branches }}
+    steps:
+      - name: setup
+        run: |
+          MANUAL_BRANCHES="$INPUT_BRANCHES"
+          [ -z "$MANUAL_BRANCHES" ] && MANUAL_BRANCHES="$DEFAULT_BRANCHES" || :
+          echo "manual_branches=${MANUAL_BRANCHES}"
+          echo "manual_branches=${MANUAL_BRANCHES}" >>${GITHUB_OUTPUT}
+          exit 0
+        
+      - name: DumpEnvironment
+        uses: asterisk/asterisk-ci-actions/DumpEnvironmentAction@main
+        with:
+          action-inputs: ${{toJSON(inputs)}}
+          action-vars: ${{ toJSON(steps.setup.outputs) }}
+
+  CreateDocsScheduledMatrix:
+    needs: [ CreateDocsDebug ]
+    if: ${{github.event_name == 'schedule' && fromJSON(vars.WIKIDOCS_ENABLE) == true }}
+    continue-on-error: false
+    strategy:
+      fail-fast: false
+      matrix:
+        branch: ${{ fromJSON(vars.WIKIDOC_BRANCHES) }}
+    runs-on: ubuntu-latest
+    steps:
+      - name: CreateDocs for ${{matrix.branch}}
+        uses: asterisk/asterisk-ci-actions/CreateAsteriskDocsComposite@main
+        with:
+          asterisk_repo:     ${{env.ASTERISK_REPO}}
+          base_branch:       ${{matrix.branch}}
+          docs_dir:          docs_dir/${{matrix.branch}}
+          github_token:      ${{secrets.GITHUB_TOKEN}}
+
+  CreateDocsScheduled:
+    needs: [ CreateDocsScheduledMatrix ]
+    if: ${{  success() || failure() }}
+    runs-on: ubuntu-latest
+    steps:
+      - name: Check CreateDocsScheduledMatrix status
+        env:
+          RESULT:    ${{needs.CreateDocsScheduledMatrix.result}}
+          GH_TOKEN:  ${{ secrets.GITHUB_TOKEN }}
+        run: |
+          case $RESULT in
+            success)
+              echo "::notice::Docs created"
+              exit 0
+              ;;
+            skipped)
+              echo "::notice::Skipped"
+              exit 1
+              ;;
+            *)
+              echo "::error::One or CreateDocs failed ($RESULT)"
+              exit 1
+          esac
+
+  CreateDocsManualMatrix:
+    needs: [ CreateDocsDebug ]
+    if: ${{github.event_name == 'workflow_dispatch'}}
+    continue-on-error: false
+    strategy:
+      fail-fast: false
+      matrix:
+        branch: ${{ fromJSON(vars.WIKIDOC_MANUAL_BRANCHES) }}
+    runs-on: ubuntu-latest
+    steps:
+      - name: CreateDocs for ${{matrix.branch}}
+        uses: asterisk/asterisk-ci-actions/CreateAsteriskDocsComposite@main
+        with:
+          asterisk_repo:     ${{env.ASTERISK_REPO}}
+          base_branch:       ${{matrix.branch}}
+          docs_dir:          docs_dir/${{matrix.branch}}
+          github_token:      ${{secrets.GITHUB_TOKEN}}
+
+  CreateDocsManual:
+    needs: [ CreateDocsManualMatrix ]
+    if: ${{  success() || failure() }}
+    runs-on: ubuntu-latest
+    steps:
+      - name: Check CreateDocsManualMatrix status
+        env:
+          RESULT:    ${{needs.CreateDocsManualMatrix.result}}
+          GH_TOKEN:  ${{ secrets.GITHUB_TOKEN }}
+        run: |
+          case $RESULT in
+            success)
+              echo "::notice::Docs created"
+              exit 0
+              ;;
+            skipped)
+              echo "::notice::Skipped"
+              exit 1
+              ;;
+            *)
+              echo "::error::One or CreateDocs failed ($RESULT)"
+              exit 1
+          esac

.github/workflows/IssueOpened.yml

@@ -0,0 +1,15 @@
+name: Issue Opened
+run-name: "Issue ${{github.event.number}} ${{github.event.action}} by ${{github.actor}}"
+on:
+  issues:
+    types: opened
+
+jobs:
+  triage:
+    runs-on: ubuntu-latest
+    steps:
+      - name: initial labeling
+        uses: andymckay/labeler@master
+        with:
+          add-labels: "triage"
+          ignore-if-labeled: true

.github/workflows/MergeApproved.yml

@@ -0,0 +1,190 @@
+name: MergeApproved
+run-name: "Merge Approved for PR ${{github.event.number}}"
+on:
+  pull_request_target:
+    types: [labeled]
+
+env:
+  PR_NUMBER:          ${{ github.event.number }}
+  BASE_BRANCH:        ${{github.event.pull_request.base.ref}}
+  MODULES_BLACKLIST:  ${{ vars.GATETEST_MODULES_BLACKLIST }} ${{ vars.UNITTEST_MODULES_BLACKLIST }}
+  FORCE:              ${{ endsWith(github.event.label.name, '-force') }}
+
+jobs:
+  IdentifyBranches:
+    if: contains(fromJSON(vars.MERGE_APPROVED_LABELS), github.event.label.name)
+    outputs:
+      branches:     ${{ steps.getbranches.outputs.branches }}
+      all_branches: ${{ steps.checkbranches.outputs.all_branches }}
+      branch_count: ${{ steps.getbranches.outputs.branch_count }}
+    runs-on: ubuntu-latest
+    steps:
+      - name: Clean up labels
+        env:
+          GH_TOKEN:          ${{ secrets.GITHUB_TOKEN }}
+        run: |
+          gh pr edit --repo ${{github.repository}} \
+            --remove-label ${{github.event.label.name}} \
+            --remove-label ${{vars.PRE_MERGE_CHECKS_PASSED_LABEL}} \
+            --remove-label ${{vars.PRE_MERGE_CHECKS_FAILED_LABEL}} \
+            --remove-label ${{vars.PRE_MERGE_GATES_PASSED_LABEL}} \
+            --remove-label ${{vars.PRE_MERGE_GATES_FAILED_LABEL}} \
+            --remove-label ${{vars.PRE_MERGE_TESTING_IN_PROGRESS}} \
+            ${{env.PR_NUMBER}} || :
+
+      - name: Get cherry-pick branches
+        uses: asterisk/asterisk-ci-actions/GetCherryPickBranchesFromPR@main
+        id: getbranches
+        with:
+          repo:                   ${{github.repository}}
+          pr_number:              ${{env.PR_NUMBER}}
+          cherry_pick_regex:      ${{vars.CHERRY_PICK_REGEX}}
+          github_token:           ${{secrets.GITHUB_TOKEN}}
+
+      - name: Check Branch Count
+        id: checkbranches
+        env:
+          BRANCH_COUNT:  ${{ steps.getbranches.outputs.branch_count }}
+          BRANCHES:  ${{ steps.getbranches.outputs.branches }}
+          GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
+        run: |
+          gh pr edit --repo ${{github.repository}} \
+          --add-label ${{vars.PRE_MERGE_TESTING_IN_PROGRESS}} \
+          ${{env.PR_NUMBER}} || :
+          all_branches=$(echo "$BRANCHES" | jq -c "[ \"$BASE_BRANCH\" ] + .")
+          echo "all_branches=${all_branches}" >>${GITHUB_OUTPUT}
+
+      - name: Pre Check Cherry-Picks
+        if: ${{ steps.getbranches.outputs.branch_count > 0 }}
+        uses: asterisk/asterisk-ci-actions/CherryPick@main
+        with:
+          repo:              ${{github.repository}}
+          pr_number:         ${{env.PR_NUMBER}}
+          branches:          ${{steps.getbranches.outputs.branches}}
+          github_token:      ${{secrets.GITHUB_TOKEN}}
+          push:              false
+
+  PreMergeUnitTestMatrix:
+    needs: [ IdentifyBranches ]
+    if: success()
+    continue-on-error: false
+    strategy:
+      fail-fast: false
+      matrix:
+        branch: ${{ fromJSON(needs.IdentifyBranches.outputs.all_branches) }}
+    runs-on: ubuntu-latest
+    steps:
+      - name: Run Unit Tests for branch ${{matrix.branch}}
+        uses: asterisk/asterisk-ci-actions/AsteriskUnitComposite@main
+        with:
+          asterisk_repo:     ${{github.repository}}
+          pr_number:         ${{env.PR_NUMBER}}
+          base_branch:       ${{matrix.branch}}
+          is_cherry_pick:    true
+          modules_blacklist: ${{env.MODULES_BLACKLIST}}
+          github_token:      ${{secrets.GITHUB_TOKEN}}
+          unittest_command:  ${{vars.UNITTEST_COMMAND}}
+
+  PreMergeUnitTests:
+    needs: [ IdentifyBranches, PreMergeUnitTestMatrix ]
+    runs-on: ubuntu-latest
+    steps:
+      - name: Check unit test matrix status
+        env:
+          RESULT:    ${{needs.PreMergeUnitTestMatrix.result}}
+          GH_TOKEN:  ${{ secrets.GITHUB_TOKEN }}
+        run: |
+          case $RESULT in
+            success)
+              gh pr edit --repo ${{github.repository}} \
+                --remove-label ${{vars.PRE_MERGE_TESTING_IN_PROGRESS}} \
+                --add-label ${{vars.PRE_MERGE_CHECKS_PASSED_LABEL}} \
+                ${{env.PR_NUMBER}} || :
+              echo "::notice::All tests passed"
+              exit 0
+              ;;
+            skipped)
+              gh pr edit --repo ${{github.repository}} \
+                --remove-label ${{vars.PRE_MERGE_TESTING_IN_PROGRESS}} \
+                --add-label ${{vars.PRE_MERGE_CHECKS_FAILED_LABEL}} \
+                ${{env.PR_NUMBER}} || :
+              echo "::notice::Unit tests were skipped because of an earlier failure"
+              exit 1
+              ;;
+            *)
+              gh pr edit --repo ${{github.repository}} \
+                --remove-label ${{vars.PRE_MERGE_TESTING_IN_PROGRESS}} \
+                --add-label ${{vars.PRE_MERGE_CHECKS_FAILED_LABEL}} \
+                ${{env.PR_NUMBER}} || :
+              echo "::error::One or more tests failed ($RESULT)"
+              exit 1
+          esac
+
+  MergeAndCherryPick:
+    needs: [ IdentifyBranches, PreMergeUnitTests ]
+    if: success()
+    concurrency:
+      group: MergeAndCherryPick
+      cancel-in-progress: false
+    runs-on: ubuntu-latest
+    steps:
+      - name: Start Merge
+        env:
+          GH_TOKEN:  ${{ secrets.GITHUB_TOKEN }}
+        run: |
+            gh pr edit --repo ${{github.repository}} \
+              --add-label ${{vars.MERGE_IN_PROGRESS_LABEL}} \
+              ${{env.PR_NUMBER}} || :
+
+      - name: Get Token needed to push cherry-picks
+        id: get_workflow_token
+        uses: peter-murray/workflow-application-token-action@v1
+        with:
+          application_id: ${{secrets.ASTERISK_ORG_ACCESS_APP_ID}}
+          application_private_key: ${{secrets.ASTERISK_ORG_ACCESS_APP_PRIV_KEY}}
+          organization: asterisk
+
+      - name: Merge and Cherry Pick to ${{needs.IdentifyBranches.outputs.branches}}
+        id: mergecp
+        uses: asterisk/asterisk-ci-actions/MergeAndCherryPickComposite@main
+        with:
+          repo:          ${{github.repository}}
+          pr_number:     ${{env.PR_NUMBER}}
+          branches:      ${{needs.IdentifyBranches.outputs.branches}}
+          force:         ${{env.FORCE}}
+          github_token:  ${{steps.get_workflow_token.outputs.token}}
+
+      - name: Merge Cleanup
+        if: always()
+        env:
+          RESULT:    ${{ steps.mergecp.outcome }}
+          GH_TOKEN:  ${{ secrets.GITHUB_TOKEN }}
+          BRANCH_COUNT: ${{ needs.IdentifyBranches.outputs.branch_count }}
+          BRANCHES:     ${{ needs.IdentifyBranches.outputs.branches }}
+
+        run: |
+          case $RESULT in
+            success)
+              gh pr edit --repo ${{github.repository}} \
+                --remove-label ${{vars.MERGE_IN_PROGRESS_LABEL}} \
+                ${{env.PR_NUMBER}} || :
+              if [ $BRANCH_COUNT -eq 0 ] ; then
+                gh pr comment --repo ${{github.repository}} \
+                  -b "Successfully merged to branch $BASE_BRANCH." \
+                  ${{env.PR_NUMBER}} || :
+              else
+                gh pr comment --repo ${{github.repository}} \
+                  -b "Successfully merged to branch $BASE_BRANCH and cherry-picked to $BRANCHES" \
+                  ${{env.PR_NUMBER}} || :
+              fi
+              exit 0
+              ;;
+            failure)
+              gh pr edit --repo ${{github.repository}} \
+                --remove-label ${{vars.MERGE_IN_PROGRESS_LABEL}} \
+                --add-label ${{vars.MERGE_FAILED_LABEL}} \
+                ${{env.PR_NUMBER}} || :
+              exit 1
+              ;;
+              *)
+          esac

.github/workflows/NightlyAdmin.yml

@@ -0,0 +1,28 @@
+name: Nightly Admin
+on:
+  schedule:
+    - cron: '30 1 * * *'
+
+env:
+  ASTERISK_REPO:     ${{ github.repository }}
+  PR_NUMBER:         0
+  PR_COMMIT:         ''
+  GITHUB_TOKEN:      ${{ secrets.GITHUB_TOKEN }}
+  GH_TOKEN:          ${{ secrets.GITHUB_TOKEN }}
+  MODULES_BLACKLIST: ${{ vars.GATETEST_MODULES_BLACKLIST }} ${{ vars.UNITTEST_MODULES_BLACKLIST }}
+
+jobs:
+  CloseStaleIssues:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Close Stale Issues
+        uses: actions/stale@v7
+        with:
+          stale-issue-message: 'This issue is stale because it has been open 7 days with no activity. Remove stale label or comment or this will be closed in 14 days.'
+          stale-issue-label: stale
+          close-issue-message: 'This issue was closed because it has been stalled for 14 days with no activity.'
+          days-before-stale: 7
+          days-before-close: 14
+          days-before-pr-close: -1
+          only-labels: triage,feedback-required
+

.github/workflows/NightlyTests.yml

@@ -0,0 +1,59 @@
+name: NightlyTests
+on:
+  workflow_dispatch:
+
+  schedule:
+    - cron: '0 2 * * *'
+
+env:
+  ASTERISK_REPO:     ${{ github.repository }}
+  PR_NUMBER:         0
+  PR_COMMIT:         ''
+  GITHUB_TOKEN:      ${{ secrets.GITHUB_TOKEN }}
+  GH_TOKEN:          ${{ secrets.GITHUB_TOKEN }}
+  MODULES_BLACKLIST: ${{ vars.GATETEST_MODULES_BLACKLIST }}
+
+jobs:
+  AsteriskNightly:
+    strategy:
+      fail-fast: false
+      matrix:
+        branch: ${{ fromJSON(vars.NIGHTLYTEST_BRANCHES) }}
+        group: ${{ fromJSON(vars.NIGHTLYTEST_LIST) }}
+    runs-on: ubuntu-latest
+    steps:
+      - name: Run Nightly Tests for ${{ matrix.group }}/${{ matrix.branch }}
+        uses: asterisk/asterisk-ci-actions/AsteriskGateComposite@main
+        with:
+          test_type:         Nightly
+          asterisk_repo:     ${{env.ASTERISK_REPO}}
+          pr_number:         ${{env.PR_NUMBER}}
+          base_branch:       ${{matrix.branch}}
+          modules_blacklist: ${{env.MODULES_BLACKLIST}}
+          github_token:      ${{secrets.GITHUB_TOKEN}}
+          testsuite_repo:    ${{vars.TESTSUITE_REPO}}
+          gatetest_group:    ${{matrix.group}}
+          gatetest_commands: ${{vars.GATETEST_COMMANDS}}
+
+  AsteriskNightlyTests:
+    if: ${{ always() }}
+    runs-on: ubuntu-latest
+    needs: AsteriskNightly
+    steps:
+      - name: Check test matrix status
+        env:
+          RESULT: ${{needs.AsteriskNightly.result}}
+        run: |
+          case $RESULT in
+            success)
+              echo "::notice::All Testsuite tests passed"
+              exit 0
+              ;;
+            skipped)
+              echo "::error::Testsuite tests were skipped because of an earlier failure"
+              exit 1
+              ;;
+              *)
+              echo "::error::One or more Testsuite tests failed"
+              exit 1
+          esac

.github/workflows/PRMerged.yml

@@ -0,0 +1,32 @@
+name: PRMerged
+run-name: "PR ${{github.event.number || inputs.pr_number}} ${{github.event.action || 'MANUAL POST MERGE'}} by ${{ github.actor }}"
+on:
+  pull_request_target:
+    types: [closed]
+  workflow_dispatch:
+    inputs:
+      pr_number:
+        description: 'PR number'
+        required: true
+        type: number
+
+concurrency:
+  group: ${{github.workflow}}-${{github.event.number || inputs.pr_number}}
+  cancel-in-progress: true
+
+env:
+  REPO:              ${{github.repository}}
+  PR_NUMBER:         ${{github.event.number || inputs.pr_number}}
+  GITHUB_TOKEN:      ${{secrets.GITHUB_TOKEN}}
+
+jobs:
+  CloseIssues:
+    if: github.event.pull_request.merged == true
+    runs-on: ubuntu-latest
+    steps:
+      - uses: wow-actions/auto-close-fixed-issues@v1
+        env:
+          GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}
+        with:
+          GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}
+

.github/workflows/PROpenedOrUpdated.yml

@@ -0,0 +1,172 @@
+name: PROpenedOrUpdated
+run-name: "PR ${{github.event.number}} ${{github.event.action}} by ${{ github.actor }}"
+on:
+#  workflow_dispatch:
+  pull_request_target:
+    types: [opened, reopened, synchronize]
+
+env:
+  ASTERISK_REPO:     ${{github.repository}}
+  PR_NUMBER:         ${{github.event.number}}
+  PR_COMMIT:         ${{github.event.pull_request.head.sha}}
+  BRANCH:            ${{github.event.pull_request.base.ref}}
+  GITHUB_TOKEN:      ${{secrets.GITHUB_TOKEN}}
+  MODULES_BLACKLIST: ${{vars.GATETEST_MODULES_BLACKLIST}} ${{vars.UNITTEST_MODULES_BLACKLIST}}
+
+jobs:
+
+  PROpenUpdateUnitTests:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Get Token needed to add reviewers
+        if: github.event.action == 'opened'
+        id: get_workflow_token
+        uses: peter-murray/workflow-application-token-action@v1
+        with:
+          application_id: ${{secrets.ASTERISK_ORG_ACCESS_APP_ID}}
+          application_private_key: ${{secrets.ASTERISK_ORG_ACCESS_APP_PRIV_KEY}}
+          organization: asterisk
+
+      - name: Get cherry-pick branches
+        uses: asterisk/asterisk-ci-actions/GetCherryPickBranchesFromPR@main
+        id: getbranches
+        with:
+          repo:                    ${{github.repository}}
+          pr_number:               ${{env.PR_NUMBER}}
+          cherry_pick_regex:       ${{vars.CHERRY_PICK_REGEX}}
+          github_token:            ${{secrets.GITHUB_TOKEN}}
+
+      - name: Add cherry-pick reminder
+        if: ${{ steps.getbranches.outputs.branch_count == 0 || steps.getbranches.outputs.forced_none != 'true' }}
+        env:
+          GITHUB_TOKEN: ${{steps.get_workflow_token.outputs.token}}
+          GH_TOKEN: ${{steps.get_workflow_token.outputs.token}}
+          CHERRY_PICK_REMINDER: ${{vars.CHERRY_PICK_REMINDER}}
+        run: |
+          IFS=$'; \n'
+          # If there's already a reminder comment, don't add another one.
+          ADD_COMMENT=true
+          # This query will FAIL if it finds the comment.
+          gh pr view --repo ${{github.repository}} --json comments \
+            --jq '.comments[].body | select(. | startswith("<!--CPR-->")) | halt_error(1)' \
+            ${{env.PR_NUMBER}} >/dev/null 2>&1 || ADD_COMMENT=false
+          if $ADD_COMMENT ; then
+            echo "Adding CPR comment"
+            gh pr comment --repo ${{github.repository}} \
+              -b "${CHERRY_PICK_REMINDER}" ${{env.PR_NUMBER}}
+          else
+            echo "CPR comment already present"
+          fi
+
+      - name: Add reviewers
+        if: github.event.action == 'opened'
+        env:
+          GITHUB_TOKEN: ${{steps.get_workflow_token.outputs.token}}
+          GH_TOKEN: ${{steps.get_workflow_token.outputs.token}}
+          CHERRY_PICK_REMINDER: ${{vars.CHERRY_PICK_REMINDER}}
+          REVIEWERS: ${{vars.PR_REVIEWERS}}
+        run: |
+          IFS=$'; \n'
+          for r in $REVIEWERS ; do
+            echo "Adding reviewer $r"
+            gh pr edit --repo ${{github.repository}} ${PR_NUMBER} --add-reviewer $r || :
+          done
+
+      - name: Set Labels
+        env:
+          GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
+        run: |
+          gh pr edit --repo ${{github.repository}} \
+            --remove-label ${{vars.TEST_CHECKS_PASSED_LABEL}} \
+            --remove-label ${{vars.TEST_CHECKS_FAILED_LABEL}} \
+            --remove-label ${{vars.TEST_GATES_PASSED_LABEL}} \
+            --remove-label ${{vars.TEST_GATES_FAILED_LABEL}} \
+            --remove-label ${{vars.CHERRY_PICK_CHECKS_PASSED_LABEL}} \
+            --remove-label ${{vars.CHERRY_PICK_CHECKS_FAILED_LABEL}} \
+            --remove-label ${{vars.CHERRY_PICK_GATES_PASSED_LABEL}} \
+            --remove-label ${{vars.CHERRY_PICK_GATES_FAILED_LABEL}} \
+            --add-label ${{vars.TESTING_IN_PROGRESS}} \
+            ${{env.PR_NUMBER}} || :
+
+      - name: Run Unit Tests
+        uses: asterisk/asterisk-ci-actions/AsteriskUnitComposite@main
+        with:
+          asterisk_repo:     ${{env.ASTERISK_REPO}}
+          pr_number:         ${{env.PR_NUMBER}}
+          base_branch:       ${{env.BRANCH}}
+          modules_blacklist: ${{env.MODULES_BLACKLIST}}
+          github_token:      ${{secrets.GITHUB_TOKEN}}
+          unittest_command:  ${{vars.UNITTEST_COMMAND}}
+
+      - name: Add Checks Passed Label
+        if: ${{ success() }}
+        env:
+          GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
+        run: |
+          gh pr edit --repo ${{github.repository}} \
+            --add-label ${{vars.TEST_CHECKS_PASSED_LABEL}} \
+            ${{env.PR_NUMBER}} || :
+
+  PROpenUpdateGateTestMatrix:
+    needs: PROpenUpdateUnitTests
+    continue-on-error: false
+    strategy:
+      fail-fast: false
+      matrix:
+        group: ${{ fromJSON(vars.GATETEST_LIST) }}
+    runs-on: ubuntu-latest
+    steps:
+      - id: runtest
+        name: Run Gate Tests for ${{ matrix.group }}
+        uses: asterisk/asterisk-ci-actions/AsteriskGateComposite@main
+        with:
+          test_type:         Gate
+          asterisk_repo:     ${{env.ASTERISK_REPO}}
+          pr_number:         ${{env.PR_NUMBER}}
+          base_branch:       ${{env.BRANCH}}
+          modules_blacklist: ${{env.MODULES_BLACKLIST}}
+          github_token:      ${{secrets.GITHUB_TOKEN}}
+          testsuite_repo:    ${{vars.TESTSUITE_REPO}}
+          gatetest_group:    ${{matrix.group}}
+          gatetest_commands: ${{vars.GATETEST_COMMANDS}}
+
+
+  PROpenUpdateGateTests:
+    if: always()
+    runs-on: ubuntu-latest
+    needs: PROpenUpdateGateTestMatrix
+    steps:
+      - name: Check test matrix status
+        env:
+          RESULT: ${{ needs.PROpenUpdateGateTestMatrix.result }}
+          GH_TOKEN: ${{ secrets.GITHUB_TOKEN }}
+        run: |
+          echo "all results: ${{ toJSON(needs.*.result) }}"
+          echo "composite result: $RESULT"
+
+          gh pr edit --repo ${{github.repository}} \
+            --remove-label ${{vars.TESTING_IN_PROGRESS}} \
+            ${{env.PR_NUMBER}} || :
+
+          case $RESULT in
+            success)
+              gh pr edit --repo ${{github.repository}} \
+                --add-label ${{vars.TEST_GATES_PASSED_LABEL}} \
+                ${{env.PR_NUMBER}} || :
+              echo "::notice::All Testsuite tests passed"
+              exit 0
+              ;;
+            skipped)
+              gh pr edit --repo ${{github.repository}} \
+                --add-label ${{vars.TEST_CHECKS_FAILED_LABEL}} \
+                ${{env.PR_NUMBER}} || :
+              echo "::error::Testsuite tests were skipped because of an earlier failure"
+              exit 1
+              ;;
+            *)
+              gh pr edit --repo ${{github.repository}} \
+                --add-label ${{vars.TEST_GATES_FAILED_LABEL}} \
+                ${{env.PR_NUMBER}} || :
+              echo "::error::One or more Testsuite tests failed ($RESULT)"
+              exit 1
+          esac

.github/workflows/Releaser.yml

@@ -0,0 +1,84 @@
+# yaml-language-server: $schema=https://json.schemastore.org/github-workflow.json
+name: Releaser
+run-name: ${{ github.actor }} is creating ${{vars.PRODUCT_NAME}} release ${{inputs.new_version}}
+on:
+  workflow_dispatch:
+    inputs:
+      is_security:
+        description: |
+          Security or Hotfix?
+        required: true
+        type: boolean
+        default: false
+      new_version:
+        description: |
+          New Version:
+          Examples:
+            20.4.0-rc1, 20.4.0-rc2, 20.4.0, 20.4.1
+            certified-20.4-cert1-rc1, certified-20.4-cert1
+        required: true
+        type: string
+      start_version:
+        description: |
+          Last Version:
+          Only use when you KNOW that the automated
+          process won't get it right.
+        required: false
+        type: string
+      push_release_branches:
+        description: |
+          Push release branches live?
+        required: true
+        type: boolean
+        default: false
+      create_github_release:
+        description: |
+          Create the GitHub release?
+        required: true
+        type: boolean
+        default: false
+      push_tarballs:
+        description: |
+          Push tarballs to downloads server?
+        required: true
+        type: boolean
+        default: false
+      send_email:
+        description: |
+          Send announcement emails?
+        required: true
+        type: boolean
+        default: false
+
+jobs:
+  ReleaseAsterisk:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Run Releaser
+        uses: asterisk/asterisk-ci-actions/ReleaserComposite@main
+        with:
+          product: ${{vars.PRODUCT_NAME}}
+          is_security: ${{inputs.is_security}}
+          new_version: ${{inputs.new_version}}
+          start_version: ${{inputs.start_version}}
+          push_release_branches: ${{inputs.push_release_branches}}
+          create_github_release: ${{inputs.create_github_release}}
+          push_tarballs: ${{inputs.push_tarballs}}
+          send_email: ${{inputs.send_email}}
+          repo: ${{github.repository}}
+          mail_list_ga: ${{vars.MAIL_LIST_GA}}
+          mail_list_rc: ${{vars.MAIL_LIST_RC}}
+          mail_list_cert_ga: ${{vars.MAIL_LIST_CERT_GA}}
+          mail_list_cert_rc: ${{vars.MAIL_LIST_CERT_RC}}
+          mail_list_sec: ${{vars.MAIL_LIST_SEC_ADV}}
+          sec_adv_url_base: ${{vars.SEC_ADV_URL_BASE}}
+          gpg_private_key: ${{secrets.ASTDEV_GPG_PRIV_KEY}}
+          github_token: ${{secrets.GITHUB_TOKEN}}
+          application_id: ${{secrets.ASTERISK_ORG_ACCESS_APP_ID}}
+          application_private_key: ${{secrets.ASTERISK_ORG_ACCESS_APP_PRIV_KEY}}
+          asteriskteamsa_username: ${{secrets.ASTERISKTEAMSA_GMAIL_ACCT}}
+          asteriskteamsa_token: ${{secrets.ASTERISKTEAMSA_GMAIL_TOKEN}}
+          deploy_ssh_priv_key: ${{secrets.DOWNLOADS_DEPLOY_SSH_PRIV_KEY}}
+          deploy_ssh_username: ${{secrets.DOWNLOADS_DEPLOY_SSH_USERNAME}}
+          deploy_host: ${{vars.DEPLOY_HOST}}
+          deploy_dir: ${{vars.DEPLOY_DIR}}

.gitignore

@@ -37,4 +37,4 @@ doxygen.log
 out/
 *.orig
 tests/CI/output
-
+.develvars

.gitreview

@@ -1,5 +1,6 @@
 [gerrit]
-defaultbranch=master
+defaultbranch=16
+basebranch=16
 #
 # Intentional padding to ensure it is possible to point a commit
 # to an alternative gerrit server/repository without breaking

CREDITS

@@ -40,7 +40,7 @@
 		rewrite of SIP transfers
 
 
-=== WISHLIST CONTRIBUTERS ===
+=== WISHLIST CONTRIBUTORS ===
 
  We'd like to thank the following for contributing to our wishlist
 

LICENSE

@@ -45,7 +45,7 @@ redistribution of Asterisk source code obtained from Digium, you
 should contact our licensing department to determine the necessary
 steps you must take. For more information on this policy, please read:
 
-http://www.digium.com/en/company/profile/trademarkpolicy.php
+https://www.sangoma.com/wp-content/uploads/Sangoma-Trademark-Policy.pdf
 
 If you have any questions regarding our licensing policy, please
 contact us:
@@ -53,7 +53,6 @@ contact us:
 +1.877.344.4861 (via telephone in the USA)
 +1.256.428.6000 (via telephone outside the USA)
 +1.256.864.0464 (via FAX inside or outside the USA)
-IAX2/pbx.digium.com (via IAX2)
 licensing@digium.com (via email)
 
 Digium, Inc.

Makefile

@@ -21,7 +21,7 @@
 #      on a single object just for that object
 # SOLINK - linker flags used only for creating dynamically loadable modules
 #          as .so files
-# DYLINK - linker flags used only for creating shared libaries
+# DYLINK - linker flags used only for creating shared libraries
 #          (.so files on Unix-type platforms, .dylib on Darwin)
 #
 # Values for ASTCFLAGS and ASTLDFLAGS can be specified in the
@@ -49,6 +49,7 @@ DESTDIR?=$(INSTALL_PATH)
 export DESTDIR
 
 export INSTALL_PATH       # Additional prefix for the following paths
+export ASTCACHEDIR
 export ASTETCDIR          # Path for config files
 export ASTVARRUNDIR
 export ASTSPOOLDIR
@@ -100,6 +101,11 @@ export TAR
 export PATCH
 export SED
 export NM
+export FIND
+export BASENAME
+export DIRNAME
+export XMLLINT
+export XMLSTARLET
 
 # makeopts is required unless the goal is just {dist{-}}clean
 ifeq ($(MAKECMDGOALS),clean)
@@ -134,7 +140,7 @@ empty:=
 space:=$(empty) $(empty)
 ASTTOPDIR:=$(subst $(space),\$(space),$(CURDIR))
 
-# Overwite config files on "make samples" or other config installation targets
+# Overwrite config files on "make samples" or other config installation targets
 OVERWRITE=y
 
 # Include debug and macro symbols in the executables (-g) and profiling info (-pg)
@@ -234,6 +240,10 @@ ifeq ($(OSARCH),OpenBSD)
   _ASTCFLAGS+=-pthread -ftrampolines
 endif
 
+ifeq ($(OSARCH),linux-uclibc)
+  AST_LIBS+=-lpthread -ldl
+endif
+
 ifeq ($(OSARCH),SunOS)
   _ASTCFLAGS+=-Wcast-align -DSOLARIS -I../include/solaris-compat -I/opt/ssl/include -I/usr/local/ssl/include -D_XPG4_2 -D__EXTENSIONS__
 endif
@@ -245,7 +255,7 @@ else
 endif
 ifneq ($(AWK),)
   ifneq ($(wildcard .version),)
-    ASTERISKVERSIONNUM:=$(shell $(AWK) -F. '{printf "%01d%02d%02d", $$1, $$2, $$3}' .version)
+    ASTERISKVERSIONNUM:=$(shell $(SED) -e 's/^certified\///' -e 's/-cert/./' .version | $(AWK) -F. '{printf "%01d%02d%02d", $$1, $$2, $$3}')
   endif
 endif
 
@@ -317,6 +327,9 @@ else
 SUBMAKE:=$(MAKE) --quiet --no-print-directory
 endif
 
+mkfile_path := $(abspath $(lastword $(MAKEFILE_LIST)))
+mkfile_dir := $(dir $(mkfile_path))
+
 # $(MAKE) is printed in several places, and we want it to be a
 # fixed size string. Define a variable whose name has also the
 # same size, so we can easily align text.
@@ -437,7 +450,7 @@ distclean: $(SUBDIRS_DIST_CLEAN) _clean
 	rm -f include/asterisk/autoconfig.h
 	rm -f include/asterisk/buildopts.h
 	rm -rf doc/api
-	rm -f doc/asterisk-ng-doxygen
+	rm -f doc/Doxyfile
 	rm -f build_tools/menuselect-deps
 
 datafiles: _all $(CORE_XMLDOC)
@@ -470,53 +483,24 @@ endif
 		$(INSTALL) -m 644 $$x "$(DESTDIR)$(ASTDATADIR)/rest-api" ; \
 	done
 
-ifeq ($(GREP),)
-else ifeq ($(GREP),:)
-else
-  XML_core_en_US = $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null))
-endif
-
+DOC_MOD_SUBDIRS := $(filter-out third-party,$(MOD_SUBDIRS))
+XML_core_en_US := $(shell build_tools/make_xml_documentation --command=print_dependencies --source-tree=. --mod-subdirs="$(DOC_MOD_SUBDIRS)")
+# core-en_US.xml is the normal documentation created with asterisk builds.
 doc/core-en_US.xml: makeopts .lastclean $(XML_core_en_US)
-	@printf "Building Documentation For: "
-	@echo "<?xml version=\"1.0\" encoding=\"UTF-8\"?>" > $@
-	@echo "<!DOCTYPE docs SYSTEM \"appdocsxml.dtd\">" >> $@
-	@echo "<?xml-stylesheet type=\"text/xsl\" href=\"appdocsxml.xslt\"?>" >> $@
-	@echo "<docs xmlns:xi=\"http://www.w3.org/2001/XInclude\">" >> $@
-	@for x in $(MOD_SUBDIRS); do \
-		printf "$$x " ; \
-		for i in `find $$x -name '*.c'`; do \
-			$(AWK) -f build_tools/get_documentation $$i >> $@ ; \
-		done ; \
-	done
-	@echo
-	@echo "</docs>" >> $@
+	@build_tools/make_xml_documentation --command=create_xml --source-tree=. --mod-subdirs="$(DOC_MOD_SUBDIRS)" \
+		--with-moduleinfo --output-file=$@
 
-ifeq ($(GREP),)
-else ifeq ($(GREP),:)
-else
-  XML_full_en_US = $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null))
-endif
-
-doc/full-en_US.xml: makeopts .lastclean $(XML_full_en_US)
+# The full-en_US.xml target is only called by the wiki documentation generation process
+# and does special post-processing in preparation for uploading to the wiki.
+# It creates full-en_US.xml but then re-creates core-en_US.xml as well.
+doc/full-en_US.xml: makeopts .lastclean $(XML_core_en_US)
 ifeq ($(PYTHON),:)
 	@echo "--------------------------------------------------------------------------"
 	@echo "---        Please install python to build full documentation           ---"
 	@echo "--------------------------------------------------------------------------"
 else
-	@printf "Building Documentation For: "
-	@echo "<?xml version=\"1.0\" encoding=\"UTF-8\"?>" > $@
-	@echo "<!DOCTYPE docs SYSTEM \"appdocsxml.dtd\">" >> $@
-	@echo "<?xml-stylesheet type=\"text/xsl\" href=\"appdocsxml.xslt\"?>" >> $@
-	@echo "<docs xmlns:xi=\"http://www.w3.org/2001/XInclude\">" >> $@
-	@for x in $(filter-out third-party,$(MOD_SUBDIRS)); do \
-		printf "$$x " ; \
-		for i in `find $$x -name '*.c'`; do \
-			$(PYTHON) build_tools/get_documentation.py < $$i >> $@ ; \
-		done ; \
-	done
-	@echo
-	@echo "</docs>" >> $@
-	@$(PYTHON) build_tools/post_process_documentation.py -i $@ -o "doc/core-en_US.xml"
+	@build_tools/make_xml_documentation --command=create_xml --source-tree=. --mod-subdirs="$(DOC_MOD_SUBDIRS)" \
+		--for-wiki --output-file=$@ --core-output-file=./doc/core-en_US.xml
 endif
 
 validate-docs: doc/core-en_US.xml
@@ -551,18 +535,17 @@ update:
 
 NEWHEADERS=$(notdir $(wildcard include/asterisk/*.h))
 OLDHEADERS=$(filter-out $(NEWHEADERS) $(notdir $(DESTDIR)$(ASTHEADERDIR)),$(notdir $(wildcard $(DESTDIR)$(ASTHEADERDIR)/*.h)))
-INSTALLDIRS="$(ASTLIBDIR)" "$(ASTMODDIR)" "$(ASTSBINDIR)" "$(ASTETCDIR)" "$(ASTVARRUNDIR)" \
+INSTALLDIRS="$(ASTLIBDIR)" "$(ASTMODDIR)" "$(ASTSBINDIR)" "$(ASTCACHEDIR)" "$(ASTETCDIR)" "$(ASTVARRUNDIR)" \
 	"$(ASTSPOOLDIR)" "$(ASTSPOOLDIR)/dictate" "$(ASTSPOOLDIR)/meetme" \
 	"$(ASTSPOOLDIR)/monitor" "$(ASTSPOOLDIR)/system" "$(ASTSPOOLDIR)/tmp" \
 	"$(ASTSPOOLDIR)/voicemail" "$(ASTSPOOLDIR)/recording" \
-	"$(ASTHEADERDIR)" "$(ASTHEADERDIR)/doxygen" \
 	"$(ASTLOGDIR)" "$(ASTLOGDIR)/cdr-csv" "$(ASTLOGDIR)/cdr-custom" \
 	"$(ASTLOGDIR)/cel-custom" "$(ASTDATADIR)" "$(ASTDATADIR)/documentation" \
 	"$(ASTDATADIR)/documentation/thirdparty" "$(ASTDATADIR)/firmware" \
 	"$(ASTDATADIR)/firmware/iax" "$(ASTDATADIR)/images" "$(ASTDATADIR)/keys" \
 	"$(ASTDATADIR)/phoneprov" "$(ASTDATADIR)/rest-api" "$(ASTDATADIR)/static-http" \
 	"$(ASTDATADIR)/sounds" "$(ASTDATADIR)/moh" "$(ASTMANDIR)/man8" "$(AGI_DIR)" "$(ASTDBDIR)" \
-	"$(ASTDATADIR)/third-party"
+	"$(ASTDATADIR)/third-party" "${ASTDATADIR}/keys/stir_shaken"
 
 installdirs:
 	@for i in $(INSTALLDIRS); do \
@@ -574,19 +557,13 @@ installdirs:
 main-bininstall:
 	+@DESTDIR="$(DESTDIR)" ASTSBINDIR="$(ASTSBINDIR)" ASTLIBDIR="$(ASTLIBDIR)" $(SUBMAKE) -C main bininstall
 
-bininstall: _all installdirs $(SUBDIRS_INSTALL) main-bininstall
+bininstall: _all installdirs $(SUBDIRS_INSTALL) main-bininstall install-headers
 	$(INSTALL) -m 755 contrib/scripts/astversion "$(DESTDIR)$(ASTSBINDIR)/"
 	$(INSTALL) -m 755 contrib/scripts/astgenkey "$(DESTDIR)$(ASTSBINDIR)/"
 	$(INSTALL) -m 755 contrib/scripts/autosupport "$(DESTDIR)$(ASTSBINDIR)/"
-	if [ ! -f /sbin/launchd ]; then \
-		./build_tools/install_subst contrib/scripts/safe_asterisk "$(DESTDIR)$(ASTSBINDIR)/safe_asterisk"; \
-	fi
-	$(INSTALL) -m 644 include/asterisk.h "$(DESTDIR)$(includedir)"
-	$(INSTALL) -m 644 include/asterisk/*.h "$(DESTDIR)$(ASTHEADERDIR)"
-	$(INSTALL) -m 644 include/asterisk/doxygen/*.h "$(DESTDIR)$(ASTHEADERDIR)/doxygen"
-	if [ -n "$(OLDHEADERS)" ]; then \
-		for h in $(OLDHEADERS); do rm -f "$(DESTDIR)$(ASTHEADERDIR)/$$h"; done \
-	fi
+ifneq ($(HAVE_SBIN_LAUNCHD),1)
+	./build_tools/install_subst contrib/scripts/safe_asterisk "$(DESTDIR)$(ASTSBINDIR)/safe_asterisk";
+endif
 
 ifneq ($(DISABLE_XMLDOC),yes)
 	$(INSTALL) -m 644 doc/core-*.xml "$(DESTDIR)$(ASTDATADIR)/documentation"
@@ -719,7 +696,17 @@ ifneq ($(filter ~%,$(DESTDIR)),)
 	@exit 1
 endif
 
-install: badshell bininstall datafiles
+versioncheck:
+ifeq ($(ASTERISKVERSION),UNKNOWN__git_check_fail)
+	@echo "Asterisk Version is unknown due to a git error. If you are running make"
+	@echo "as a different user than the project owner, this can be resolved by"
+	@echo "running the following command as the user currently executing make: "$$USER
+	@echo "git config --global --add safe.directory "$(mkfile_dir:/=)
+	@exit 1
+endif
+
+
+install: badshell versioncheck bininstall datafiles
 	@if [ -x /usr/sbin/asterisk-post-install ]; then \
 		/usr/sbin/asterisk-post-install "$(DESTDIR)" . ; \
 	fi
@@ -763,7 +750,7 @@ upgrade: bininstall
 #  (1) the configuration directory to install from
 #  (2) the extension to strip off
 define INSTALL_CONFIGS
-	@for x in configs/$(1)/*$(2); do \
+	@for x in $(1)/*$(2); do \
 		dst="$(DESTDIR)$(ASTETCDIR)/`$(BASENAME) $$x $(2)`"; \
 		if [ -f "$${dst}" ]; then \
 			if [ "$(OVERWRITE)" = "y" ]; then \
@@ -782,7 +769,8 @@ define INSTALL_CONFIGS
 	done ; \
 	if [ "$(OVERWRITE)" = "y" ]; then \
 		echo "Updating asterisk.conf" ; \
-		sed -e 's|^astetcdir.*$$|astetcdir => $(ASTETCDIR)|' \
+		sed -e 's|^astcachedir.*$$|astcachedir => $(ASTCACHEDIR)|' \
+			-e 's|^astetcdir.*$$|astetcdir => $(ASTETCDIR)|' \
 			-e 's|^astmoddir.*$$|astmoddir => $(ASTMODDIR)|' \
 			-e 's|^astvarlibdir.*$$|astvarlibdir => $(ASTVARLIBDIR)|' \
 			-e 's|^astdbdir.*$$|astdbdir => $(ASTDBDIR)|' \
@@ -799,6 +787,24 @@ define INSTALL_CONFIGS
 	fi
 endef
 
+install-headers:
+# Will create all components of "$(DESTDIR)$(ASTHEADERDIR)/doxygen" including "$(DESTDIR)$(includedir)"
+	$(INSTALL) -d "$(DESTDIR)$(ASTHEADERDIR)/doxygen"
+	$(INSTALL) -m 644 include/asterisk.h "$(DESTDIR)$(includedir)"
+	$(INSTALL) -m 644 include/asterisk/*.h "$(DESTDIR)$(ASTHEADERDIR)"
+	$(INSTALL) -m 644 include/asterisk/doxygen/*.h "$(DESTDIR)$(ASTHEADERDIR)/doxygen"
+	if [ -n "$(OLDHEADERS)" ]; then \
+		for h in $(OLDHEADERS); do rm -f "$(DESTDIR)$(ASTHEADERDIR)/$$h"; done \
+	fi
+
+install-configs:
+	@if test -z "$(CONFIG_SRC)" -o ! -d "$(CONFIG_SRC)"; then \
+		>&2 echo "CONFIG_SRC must be set to a directory."; \
+		exit 1; \
+	fi
+	@echo "Installing config files from $(CONFIG_SRC)/*$(CONFIG_EXTEN)"
+	$(call INSTALL_CONFIGS,$(CONFIG_SRC),$(CONFIG_EXTEN))
+
 # XXX why *.adsi is installed first ?
 adsi:
 	@echo Installing adsi config files...
@@ -815,7 +821,7 @@ adsi:
 
 samples: adsi
 	@echo Installing other config files...
-	$(call INSTALL_CONFIGS,samples,.sample)
+	$(call INSTALL_CONFIGS,configs/samples,.sample)
 	$(INSTALL) -d "$(DESTDIR)$(ASTSPOOLDIR)/voicemail/default/1234/INBOX"
 	build_tools/make_sample_voicemail "$(DESTDIR)/$(ASTDATADIR)" "$(DESTDIR)/$(ASTSPOOLDIR)"
 	@for x in phoneprov/*; do \
@@ -838,7 +844,7 @@ samples: adsi
 
 basic-pbx:
 	@echo Installing basic-pbx config files...
-	$(call INSTALL_CONFIGS,basic-pbx)
+	$(call INSTALL_CONFIGS,configs/basic-pbx)
 
 webvmail:
 	@[ -d "$(DESTDIR)$(HTTP_DOCSDIR)/" ] || ( printf "http docs directory not found.\nUpdate assignment of variable HTTP_DOCSDIR in Makefile!\n" && exit 1 )
@@ -867,29 +873,36 @@ webvmail:
 	@echo " +-------------------------------------------+"
 
 progdocs:
-# Note, Makefile conditionals must not be tabbed out. Wasted hours with that.
-	@cp doc/asterisk-ng-doxygen.in doc/asterisk-ng-doxygen
 ifeq ($(DOXYGEN),:)
 	@echo "Doxygen is not installed.  Please install and re-run the configuration script."
 else
+	@cp doc/Doxyfile.in doc/Doxyfile
 ifeq ($(DOT),:)
 	@echo "DOT is not installed. Doxygen will not produce any diagrams. Please install and re-run the configuration script."
 else
-	# Enable DOT
-	@echo "HAVE_DOT = YES" >> doc/asterisk-ng-doxygen
+	@echo "HAVE_DOT = YES" >> doc/Doxyfile
 endif
-	# Set Doxygen PROJECT_NUMBER variable
-ifneq ($(ASTERISKVERSION),UNKNOWN__and_probably_unsupported)
-	@echo "PROJECT_NUMBER = $(ASTERISKVERSION)" >> doc/asterisk-ng-doxygen
+ifneq ($(NOISY_BUILD),yes)
+	@echo "EXTRACT_ALL = YES" >> doc/Doxyfile
+endif
+ifeq ($(AST_DEVMODE),yes)
+	@echo "INTERNAL_DOCS = YES" >> doc/Doxyfile
+	@echo "WARN_NO_PARAMDOC = YES" >> doc/Doxyfile
+endif
+ifeq ($(ASTERISKVERSION),UNKNOWN__and_probably_unsupported)
+	@echo "Asterisk Version is unknown, not configuring Doxygen PROJECT_NUMBER."
+else ifeq ($(ASTERISKVERSION),UNKNOWN__git_check_fail)
+	@echo "Asterisk Version is unknown due to a git error. If you are running make"
+	@echo "as a different user than the project owner, this can be resolved by"
+	@echo "running the following command as the user currently executing make: "$$USER
+	@echo "git config --global --add safe.directory "$(mkfile_dir:/=)
+	@echo "not configuring Doxygen PROJECT_NUMBER."
 else
-	echo "Asterisk Version is unknown, not configuring Doxygen PROJECT_NUMBER."
+	@echo "PROJECT_NUMBER = $(ASTERISKVERSION)" >> doc/Doxyfile
 endif
-	# Validate and auto-update local copy
-	@doxygen -u doc/asterisk-ng-doxygen
-	# Run Doxygen
-	@doxygen doc/asterisk-ng-doxygen
-	# Remove configuration backup file
-	@rm -f doc/asterisk-ng-doxygen.bak
+	@echo "Generating C-API documentation. This will take a while."
+	@doxygen doc/Doxyfile
+	@echo "Generation complete. Any warnings are in ./doxygen.log."
 endif
 
 install-logrotate:
@@ -938,7 +951,7 @@ config:
 		if [ -z "$(DESTDIR)" ] ; then \
 			/sbin/chkconfig --add asterisk ; \
 		fi ; \
-	elif [ -f /etc/os-release ] && [ "opensuse" = "$(shell . /etc/os-release && echo $$ID)" ] ; then \
+	elif [ -f /etc/os-release ] && [ "opensuse" = "$(shell . /etc/os-release 2>/dev/null && echo $$ID)" ] ; then \
 		./build_tools/install_subst contrib/init.d/rc.suse.asterisk  "$(DESTDIR)/etc/init.d/asterisk"; \
 		if [ ! -f /etc/sysconfig/asterisk ] ; then \
 			$(INSTALL) -m 644 contrib/init.d/etc_default_asterisk "$(DESTDIR)/etc/sysconfig/asterisk" ; \
@@ -950,7 +963,7 @@ config:
 		./build_tools/install_subst contrib/init.d/rc.archlinux.asterisk  "$(DESTDIR)/etc/init.d/asterisk"; \
 	elif [ -f /etc/slackware-version ]; then \
 		./build_tools/install_subst contrib/init.d/rc.slackware.asterisk  "$(DESTDIR)/etc/rc.d/rc.asterisk"; \
-	elif [ -f /etc/os-release ] && [ "slackware" = "$(shell . /etc/os-release && echo $$ID)" ] ; then \
+	elif [ -f /etc/os-release ] && [ "slackware" = "$(shell . /etc/os-release 2>/dev/null && echo $$ID)" ] ; then \
 		./build_tools/install_subst contrib/init.d/rc.slackware.asterisk  "$(DESTDIR)/etc/rc.d/rc.asterisk"; \
 	elif [ -d "$(DESTDIR)/Library/LaunchDaemons" ]; then \
 		if [ ! -f "$(DESTDIR)/Library/LaunchDaemons/org.asterisk.asterisk.plist" ]; then \
@@ -974,18 +987,22 @@ sounds:
 	@$(MAKE) clean
 	@[ -f "$(DESTDIR)$(ASTDBDIR)/astdb.sqlite3" ] || [ ! -f "$(DESTDIR)$(ASTDBDIR)/astdb" ] || [ ! -f menuselect.makeopts ] || grep -q MENUSELECT_UTILS=.*astdb2sqlite3 menuselect.makeopts || (sed -i.orig -e's/MENUSELECT_UTILS=\(.*\)/MENUSELECT_UTILS=\1 astdb2sqlite3/' menuselect.makeopts && echo "Updating menuselect.makeopts to include astdb2sqlite3" && echo "Original version backed up to menuselect.makeopts.orig")
 
+
 $(SUBDIRS_UNINSTALL):
 	+@DESTDIR="$(DESTDIR)" ASTSBINDIR="$(ASTSBINDIR)" ASTDATADIR="$(ASTDATADIR)" $(SUBMAKE) -C $(@:-uninstall=) uninstall
 
 main-binuninstall:
 	+@DESTDIR="$(DESTDIR)" ASTSBINDIR="$(ASTSBINDIR)" ASTLIBDIR="$(ASTLIBDIR)" $(SUBMAKE) -C main binuninstall
 
-_uninstall: $(SUBDIRS_UNINSTALL) main-binuninstall
+uninstall-headers:
+	rm -rf "$(DESTDIR)$(ASTHEADERDIR)"
+	rm -f "$(DESTDIR)$(includedir)/asterisk.h"
+
+_uninstall: $(SUBDIRS_UNINSTALL) main-binuninstall uninstall-headers
 	rm -f "$(DESTDIR)$(ASTMODDIR)/"*
 	test -n "$(_oldmoddir)" -a -d "$(_oldmoddir)" && rm -f "$(_oldmoddir)/"* || :
 	rm -f "$(DESTDIR)$(ASTSBINDIR)/astgenkey"
 	rm -f "$(DESTDIR)$(ASTSBINDIR)/autosupport"
-	rm -rf "$(DESTDIR)$(ASTHEADERDIR)"
 	rm -rf "$(DESTDIR)$(ASTDATADIR)/firmware"
 	rm -f "$(DESTDIR)$(ASTMANDIR)/man8/asterisk.8"
 	rm -f "$(DESTDIR)$(ASTMANDIR)/man8/astgenkey.8"
@@ -1015,7 +1032,7 @@ uninstall: _uninstall
 	@echo " +            $(mK) uninstall-all            +"
 	@echo " +-------------------------------------------+"
 
-uninstall-all: _uninstall
+uninstall-all: _uninstall uninstall-headers
 	rm -rf "$(DESTDIR)$(ASTMODDIR)"
 	test -n "$(_oldmoddir)" -a -d "$(_oldmoddir)" && rm -rf "$(_oldmoddir)" || :
 	rm -rf "$(DESTDIR)$(ASTVARLIBDIR)"
@@ -1101,13 +1118,16 @@ ifeq ($(PYTHON),:)
 else
 	@$(INSTALL) -d doc/rest-api
 	$(PYTHON) rest-api-templates/make_ari_stubs.py \
-		rest-api/resources.json .
+		--resources rest-api/resources.json --source-dir $(ASTTOPDIR) \
+		--dest-dir $(ASTTOPDIR)/doc/rest-api --docs-prefix ../
 endif
 
 check-alembic: makeopts
 	@find contrib/ast-db-manage/ -name '*.pyc' -delete
 	@ALEMBIC=$(ALEMBIC) build_tools/make_check_alembic config cdr voicemail >&2
 
+.PHONY: install-configs
+.PHONY: install-headers
 .PHONY: menuselect
 .PHONY: main
 .PHONY: sounds
@@ -1121,8 +1141,10 @@ check-alembic: makeopts
 .PHONY: uninstall
 .PHONY: _uninstall
 .PHONY: uninstall-all
+.PHONY: uninstall-headers
 .PHONY: badshell
 .PHONY: installdirs
+.PHONY: progdocs
 .PHONY: validate-docs
 .PHONY: _clean
 .PHONY: ari-stubs

Makefile.moddir_rules

@@ -114,7 +114,7 @@ ifneq ($(findstring :,$(XMLSTARLET)$(BASH)),:)
 		EXTERNAL_MODS=$$($(XMLSTARLET) sel -t -m "/category/member[support_level = 'external']" -v "@name" -n .moduleinfo) ;\
 		for x in $${EXTERNAL_MODS} ; do \
 			if [ -z "$${DISABLED_MODS[$${x}]}" ] ; then \
-				$(ASTTOPDIR)/build_tools/download_externals $${x} ;\
+				PJPROJECT_BUNDLED=${PJPROJECT_BUNDLED} $(ASTTOPDIR)/build_tools/download_externals $${x} ;\
 			fi ;\
 		done ;\
 	fi

Makefile.rules

@@ -20,6 +20,9 @@
 # Helpful functions
 # call with $(call function,...)
 tolower = $(shell echo $(1) | tr '[:upper:]' '[:lower:]')
+# Takes a list of MENUSELECT_CFLAG Id and returns CFLAGS to declare
+# the ones which are enabled.
+get_menuselect_cflags=$(patsubst %,-D%,$(filter $1,$(MENUSELECT_CFLAGS)))
 
 .PHONY: dist-clean
 
@@ -60,6 +63,7 @@ else
 endif
 
 OPTIMIZE?=-O3
+
 ifneq ($(findstring darwin,$(OSARCH)),)
   ifeq ($(shell if test `/usr/bin/sw_vers -productVersion | cut -c4` -gt 5; then echo 6; else echo 0; fi),6)
     # Snow Leopard/Lion has an issue with this optimization flag on large files (like chan_sip)
@@ -67,6 +71,17 @@ ifneq ($(findstring darwin,$(OSARCH)),)
   endif
 endif
 
+ifeq ($(CC),gcc)
+    # gcc version 8.2.1 and above must have partial-inlining disabled in order
+    # to avoid a documented bug. Sort to make the lowest version number come
+    # first. If it's the specified version then the current gcc version is equal
+    # to or greater, so add the custom optimization rule.
+    gcc_versions=$(shell printf "%s\n" $$(gcc -dumpversion) 8.2.1 | sort -n)
+    ifeq ($(firstword $(gcc_versions)),8.2.1)
+        OPTIMIZE+=-fno-partial-inlining
+	endif
+endif
+
 ifeq ($(findstring DONT_OPTIMIZE,$(MENUSELECT_CFLAGS))$(AST_CODE_COVERAGE),no)
     _ASTCFLAGS+=$(OPTIMIZE)
 else
@@ -189,4 +204,19 @@ endif
 	$(ECHO_PREFIX) echo "   [LD] $^ -> $@"
 	$(CMD_PREFIX) $(CXX) -o $@ $(PTHREAD_CFLAGS) $(_ASTLDFLAGS) $^ $(CXX_LIBS) $(ASTLDFLAGS)
 
+# These CC commands just create an object file with the input file embedded in it.
+# It can be access from code as follows:
+# If your input file is named abc_def.xml...
+#
+# extern const uint8_t _binary_abc_def_xml_start[];
+# extern const uint8_t _binary_abc_def_xml_end[];
+# extern const size_t _binary_abc_def_xml_size;
+%.o: %.xml
+	$(ECHO_PREFIX) echo "   [LD] $^ -> $@"
+	$(CMD_PREFIX) $(CC) -g -nostartfiles  -nodefaultlibs -nostdlib -r -Wl,-b,binary -o $@ $^
+
+%.o: %.xslt
+	$(ECHO_PREFIX) echo "   [LD] $^ -> $@"
+	$(CMD_PREFIX) $(CC) -g -nostartfiles  -nodefaultlibs -nostdlib -r -Wl,-b,binary -o $@ $^
+
 dist-clean:: clean

README-SERIOUSLY.bestpractices.md

@@ -332,7 +332,7 @@ Data: echo hello world!
 ```
 
 This manager command will attempt to execute an Asterisk application, System,
-which is normally associated with the "system" class authorication.  While some
+which is normally associated with the "system" class authorization.  While some
 checks have been put into Asterisk to take this into account, certain dialplan
 configurations and/or clever manipulation of the Originate manager action can
 circumvent these checks.  For example, take the following dialplan:
@@ -365,7 +365,7 @@ set channel variables; which allows the execution of dialplan functions.
 
 Dialplan functions within Asterisk are incredibly powerful, which is wonderful
 for building applications using Asterisk. But during the read or write
-execution, certain diaplan functions do much more. For example, reading the
+execution, certain dialplan functions do much more. For example, reading the
 SHELL() function can execute arbitrary commands on the system Asterisk is
 running on. Writing to the FILE() function can change any file that Asterisk has
 write access to.

README.md

@@ -1,7 +1,7 @@
 # The Asterisk(R) Open Source PBX
 ```text
         By Mark Spencer <markster@digium.com> and the Asterisk.org developer community.
-        Copyright (C) 2001-2016 Digium, Inc. and other copyright holders.
+        Copyright (C) 2001-2021 Sangoma Technologies Corporation and other copyright holders.
 ```
 ## SECURITY
 
@@ -47,7 +47,7 @@ and the BSD variants.
 ANY special hardware, not even a sound card) to install and run Asterisk.
 
 Supported telephony hardware includes:
-* All Analog and Digital Interface cards from [Digium]
+* All Analog and Digital Interface cards from [Sangoma]
 * QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
 * any full duplex sound card supported by ALSA, OSS, or PortAudio
 * any ISDN card supported by mISDN on Linux
@@ -91,7 +91,10 @@ guides in the [configs] directory.
 2. Run `./configure`
 
   Execute the configure script to guess values for system-dependent
-variables used during compilation.
+variables used during compilation. If the script indicates that some required 
+components are missing, you can run `./contrib/scripts/install_prereq install`
+to install the necessary components. Note that this will install all dependencies for every functionality of Asterisk. After running the script, you will need
+to rerun `./configure`.
 
 3. Run `make menuselect` _\[optional]_
 
@@ -251,7 +254,7 @@ Welcome to the growing worldwide community of Asterisk users!
 
 ---
 
-Asterisk is a trademark of Digium, Inc.
+Asterisk is a trademark of Sangoma Technologies Corporation
 
 [home page]: https://www.asterisk.org
 [support]: https://www.asterisk.org/support
@@ -261,7 +264,7 @@ Asterisk is a trademark of Digium, Inc.
 [voip-info.org]: http://www.voip-info.org/wiki-Asterisk
 [asteriskdocs.org]: http://www.asteriskdocs.org
 [NTP]: http://www.ntp.org/
-[Digium]: https://www.digium.com/
+[Sangoma]: https://www.sangoma.com/
 [UPGRADE.txt]: UPGRADE.txt
 [CHANGES]: CHANGES
 [configs]: configs

UPGRADE-1.2.txt

@@ -1,218 +0,0 @@
-=========================================================
-===
-=== Information for upgrading from Asterisk 1.0 to 1.2
-===
-=== This file documents all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also includes advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=========================================================
-
-Compiling:
-
-* The Asterisk 1.2 source code now uses C language features
-  supported only by 'modern' C compilers.  Generally, this means GCC
-  version 3.0 or higher, although some GCC 2.96 releases will also
-  work.  Some non-GCC compilers that support C99 and the common GCC
-  extensions (including anonymous structures and unions) will also
-  work.  All releases of GCC 2.95 do _not_ have the requisite feature
-  support; systems using that compiler will need to be upgraded to
-  a more recent compiler release.
-
-Dialplan Expressions:
-
-* The dialplan expression parser (which handles $[ ... ] constructs)
-  has gone through a major upgrade, but has one incompatible change:
-  spaces are no longer required around expression operators, including
-  string comparisons. However, you can now use quoting to keep strings
-  together for comparison. For more details, please read the
-  doc/README.variables file, and check over your dialplan for possible
-  problems.
-
-Agents:
-
-* The default for ackcall has been changed to "no" instead of "yes"
-  because of a bug which caused the "yes" behavior to generally act like
-  "no".  You may need to adjust the value if your agents behave
-  differently than you expect with respect to acknowledgement.
-
-* The AgentCallBackLogin application now requires a second '|' before
-  specifying an extension@context.  This is to distinguish the options
-  string from the extension, so that they do not conflict.  See
-  'show application AgentCallbackLogin' for more details.
-
-Parking:
-
-* Parking behavior has changed slightly; when a parked call times out,
-  Asterisk will attempt to deliver the call back to the extension that
-  parked it, rather than the 's' extension. If that extension is busy
-  or unavailable, the parked call will be lost.
-
-Dialing:
-
-* The Caller*ID of the outbound leg is now the extension that was
-  called, rather than the Caller*ID of the inbound leg of the call.  The
-  "o" flag for Dial can be used to restore the original behavior if
-  desired.  Note that if you are looking for the originating callerid
-  from the manager event, there is a new manager event "Dial" which
-  provides the source and destination channels and callerid.
-
-IAX:
-
-* The naming convention for IAX channels has changed in two ways:
-   1. The call number follows a "-" rather than a "/" character.
-   2. The name of the channel has been simplified to IAX2/peer-callno,
-   rather than IAX2/peer@peer-callno or even IAX2/peer@peer/callno.
-
-SIP:
-
-* The global option "port" in 1.0.X that is used to set which port to
-  bind to has been changed to "bindport" to be more consistent with
-  the other channel drivers and to avoid confusion with the "port"
-  option for users/peers.
-
-* The "Registry" event now uses "Username" rather than "User" for
-  consistency with IAX.
-
-Applications:
-
-* With the addition of dialplan functions (which operate similarly
-  to variables), the SetVar application has been renamed to Set.
-
-* The CallerPres application has been removed.  Use SetCallerPres
-  instead.  It accepts both numeric and symbolic names.
-
-* The applications GetGroupCount, GetGroupMatchCount, SetGroup, and
-  CheckGroup have been deprecated in favor of functions.  Here is a
-  table of their replacements:
-
-  GetGroupCount([groupname][@category]	       GROUP_COUNT([groupname][@category])	Set(GROUPCOUNT=${GROUP_COUNT()})
-  GroupMatchCount(groupmatch[@category])       GROUP_MATCH_COUNT(groupmatch[@category])	Set(GROUPCOUNT=${GROUP_MATCH_COUNT(SIP/.*)})
-  SetGroup(groupname[@category])	       GROUP([category])=groupname		Set(GROUP()=test)
-  CheckGroup(max[@category])		       N/A					GotoIf($[ ${GROUP_COUNT()} > 5 ]?103)
-
-  Note that CheckGroup does not have a direct replacement.  There is
-  also a new function called GROUP_LIST() which will return a space
-  separated list of all of the groups set on a channel.  The GROUP()
-  function can also return the name of the group set on a channel when
-  used in a read environment.
-
-* The applications DBGet and DBPut have been deprecated in favor of
-  functions.  Here is a table of their replacements:
-
-  DBGet(foo=family/key)        Set(foo=${DB(family/key)})
-  DBPut(family/key=${foo})     Set(DB(family/key)=${foo})
-
-* The application SetLanguage has been deprecated in favor of the
-  function LANGUAGE().
-
-  SetLanguage(fr)		Set(LANGUAGE()=fr)
-
-  The LANGUAGE function can also return the currently set language:
-
-  Set(MYLANG=${LANGUAGE()})
-
-* The applications AbsoluteTimeout, DigitTimeout, and ResponseTimeout
-  have been deprecated in favor of the function TIMEOUT(timeouttype):
-
-  AbsoluteTimeout(300)		Set(TIMEOUT(absolute)=300)
-  DigitTimeout(15)		Set(TIMEOUT(digit)=15)
-  ResponseTimeout(15)		Set(TIMEOUT(response)=15)
-
-  The TIMEOUT() function can also return the currently set timeouts:
-
-  Set(DTIMEOUT=${TIMEOUT(digit)})
-
-* The applications SetCIDName, SetCIDNum, and SetRDNIS have been
-  deprecated in favor of the CALLERID(datatype) function:
-
-  SetCIDName(Joe Cool)		Set(CALLERID(name)=Joe Cool)
-  SetCIDNum(2025551212)		Set(CALLERID(number)=2025551212)
-  SetRDNIS(2024561414)		Set(CALLERID(RDNIS)=2024561414)
-
-* The application Record now uses the period to separate the filename
-  from the format, rather than the colon.
-
-* The application VoiceMail now supports a 'temporary' greeting for each
-  mailbox. This greeting can be recorded by using option 4 in the
-  'mailbox options' menu, and 'change your password' option has been
-  moved to option 5.
-
-* The application VoiceMailMain now only matches the 'default' context if
-  none is specified in the arguments.  (This was the previously
-  documented behavior, however, we didn't follow that behavior.)  The old
-  behavior can be restored by setting searchcontexts=yes in voicemail.conf.
-
-Queues:
-
-* A queue is now considered empty not only if there are no members but if
-  none of the members are available (e.g. agents not logged on).  To
-  restore the original behavior, use "leavewhenempty=strict" or
-  "joinwhenempty=strict" instead of "=yes" for those options.
-
-* It is now possible to use multi-digit extensions in the exit context
-  for a queue (although you should not have overlapping extensions,
-  as there is no digit timeout). This means that the EXITWITHKEY event
-  in queue_log can now contain a key field with more than a single
-  character in it.
-
-Extensions:
-
-* By default, there is a new option called "autofallthrough" in
-  extensions.conf that is set to yes.  Asterisk 1.0 (and earlier)
-  behavior was to wait for an extension to be dialed after there were no
-  more extensions to execute.  "autofallthrough" changes this behavior
-  so that the call will immediately be terminated with BUSY,
-  CONGESTION, or HANGUP based on Asterisk's best guess.  If you are
-  writing an extension for IVR, you must use the WaitExten application
-  if "autofallthrough" is set to yes.
-
-AGI:
-
-* AGI scripts did not always get SIGHUP at the end, previously.  That
-  behavior has been fixed.  If you do not want your script to terminate
-  at the end of AGI being called (e.g. on a hangup) then set SIGHUP to
-  be ignored within your application.
-
-* CallerID is reported with agi_callerid and agi_calleridname instead
-  of a single parameter holding both.
-
-Music On Hold:
-
-* The preferred format for musiconhold.conf has changed; please see the
-  sample configuration file for the new format. The existing format
-  is still supported but will generate warnings when the module is loaded.
-
-chan_modem:
-
-* All the chan_modem channel drivers (aopen, bestdata and i4l) are deprecated
-  in this release, and will be removed in the next major Asterisk release.
-  Please migrate to chan_misdn for ISDN interfaces; there is no upgrade
-  path for aopen and bestdata modem users.
-
-MeetMe:
-
-* The conference application now allows users to increase/decrease their
-  speaking volume and listening volume (independently of each other and
-  other users); the 'admin' and 'user' menus have changed, and new sound
-  files are included with this release. However, if a user calling in
-  over a Zaptel channel that does NOT have hardware DTMF detection
-  increases their speaking volume, it is likely they will no longer be
-  able to enter/exit the menu or make any further adjustments, as the
-  software DTMF detector will not be able to recognize the DTMF coming
-  from their device.
-
-GetVar Manager Action:
-
-* Previously, the behavior of the GetVar manager action reported the value
-  of a variable in the following manner:
-   > name: value
-  This has been changed to a manner similar to the SetVar action and is now
-   > Variable: name
-   > Value: value

UPGRADE-1.4.txt

@@ -1,497 +0,0 @@
-=========================================================
-===
-=== Information for upgrading from Asterisk 1.2 to 1.4
-===
-=== These files document all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also includes advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
-===
-=========================================================
-
-Build Process (configure script):
-
-Asterisk now uses an autoconf-generated configuration script to learn how it
-should build itself for your system. As it is a standard script, running:
-
-$ ./configure --help
-
-will show you all the options available. This script can be used to tell the
-build process what libraries you have on your system (if it cannot find them
-automatically), which libraries you wish to have ignored even though they may
-be present, etc.
-
-You must run the configure script before Asterisk will build, although it will
-attempt to automatically run it for you with no options specified; for most
-users, that will result in a similar build to what they would have had before
-the configure script was added to the build process (except for having to run
-'make' again after the configure script is run). Note that the configure script
-does NOT need to be re-run just to rebuild Asterisk; you only need to re-run it
-when your system configuration changes or you wish to build Asterisk with
-different options.
-
-Build Process (module selection):
-
-The Asterisk source tree now includes a basic module selection and build option
-selection tool called 'menuselect'. Run 'make menuselect' to make your choices.
-In this tool, you can disable building of modules that you don't care about,
-turn on/off global options for the build and see which modules will not
-(and cannot) be built because your system does not have the required external
-dependencies installed.
-
-The resulting file from menuselect is called 'menuselect.makeopts'. Note that
-the resulting menuselect.makeopts file generally contains which modules *not*
-to build. The modules listed in this file indicate which modules have unmet
-dependencies, a present conflict, or have been disabled by the user in the
-menuselect interface. Compiler Flags can also be set in the menuselect
-interface.  In this case, the resulting file contains which CFLAGS are in use,
-not which ones are not in use.
-
-If you would like to save your choices and have them applied against all
-builds, the file can be copied to '~/.asterisk.makeopts' or
-'/etc/asterisk.makeopts'.
-
-Build Process (Makefile targets):
-
-The 'valgrind' and 'dont-optimize' targets have been removed; their functionality
-is available by enabling the DONT_OPTIMIZE setting in the 'Compiler Flags' menu
-in the menuselect tool.
-
-It is now possible to run most make targets against a single subdirectory; from
-the top level directory, for example, 'make channels' will run 'make all' in the
-'channels' subdirectory. This also is true for 'clean', 'distclean' and 'depend'.
-
-Sound (prompt) and Music On Hold files:
-
-Beginning with Asterisk 1.4, the sound files and music on hold files supplied for
-use with Asterisk have been replaced with new versions produced from high quality
-master recordings, and are available in three languages (English, French and
-Spanish) and in five formats (WAV (uncompressed), mu-Law, a-Law, GSM and G.729).
-In addition, the music on hold files provided by opsound.org Music are now available
-in the same five formats, but no longer available in MP3 format.
-
-The Asterisk 1.4 tarball packages will only include English prompts in GSM format,
-(as were supplied with previous releases) and the opsound.org MOH files in WAV format.
-All of the other variations can be installed by running 'make menuselect' and
-selecting the packages you wish to install; when you run 'make install', those
-packages will be downloaded and installed along with the standard files included
-in the tarball.
-
-If for some reason you expect to not have Internet access at the time you will be
-running 'make install', you can make your package selections using menuselect and
-then run 'make sounds' to download (only) the sound packages; this will leave the
-sound packages in the 'sounds' subdirectory to be used later during installation.
-
-WARNING: Asterisk 1.4 supports a new layout for sound files in multiple languages;
-instead of the alternate-language files being stored in subdirectories underneath
-the existing files (for French, that would be digits/fr, letters/fr, phonetic/fr,
-etc.) the new layout creates one directory under /var/lib/asterisk/sounds for the
-language itself, then places all the sound files for that language under that
-directory and its subdirectories. This is the layout that will be created if you
-select non-English languages to be installed via menuselect, HOWEVER Asterisk does
-not default to this layout and will not find the files in the places it expects them
-to be. If you wish to use this layout, make sure you put 'languageprefix=yes' in your
-/etc/asterisk/asterisk.conf file, so that Asterisk will know how the files were
-installed.
-
-PBX Core:
-
-* The (very old and undocumented) ability to use BYEXTENSION for dialing
-  instead of ${EXTEN} has been removed.
-
-* Builtin (res_features) transfer functionality attempts to use the context
-  defined in TRANSFER_CONTEXT variable of the transferer channel first. If
-  not set, it uses the transferee variable. If not set in any channel, it will
-  attempt to use the last non macro context. If not possible, it will default
-  to the current context.
-
-* The autofallthrough setting introduced in Asterisk 1.2 now defaults to 'yes';
-  if your dialplan relies on the ability to 'run off the end' of an extension
-  and wait for a new extension without using WaitExten() to accomplish that,
-  you will need set autofallthrough to 'no' in your extensions.conf file.
-
-Command Line Interface:
-
-* 'show channels concise', designed to be used by applications that will parse
-  its output, previously used ':' characters to separate fields. However, some
-  of those fields can easily contain that character, making the output not
-  parseable. The delimiter has been changed to '!'.
-
-Applications:
-
-* In previous Asterisk releases, many applications would jump to priority n+101
-  to indicate some kind of status or error condition.  This functionality was
-  marked deprecated in Asterisk 1.2.  An option to disable it was provided with
-  the default value set to 'on'.  The default value for the global priority
-  jumping option is now 'off'.
-
-* The applications Cut, Sort, DBGet, DBPut, SetCIDNum, SetCIDName, SetRDNIS,
-  AbsoluteTimeout, DigitTimeout, ResponseTimeout, SetLanguage, GetGroupCount,
-  and GetGroupMatchCount were all deprecated in version 1.2, and therefore have
-  been removed in this version.  You should use the equivalent dialplan
-  function in places where you have previously used one of these applications.
-
-* The application SetGlobalVar has been deprecated.  You should replace uses
-  of this application with the following combination of Set and GLOBAL():
-  Set(GLOBAL(name)=value).  You may also access global variables exclusively by
-  using the GLOBAL() dialplan function, instead of relying on variable
-  interpolation falling back to globals when no channel variable is set.
-
-* The application SetVar has been renamed to Set.  The syntax SetVar was marked
-  deprecated in version 1.2 and is no longer recognized in this version.  The
-  use of Set with multiple argument pairs has also been deprecated.  Please
-  separate each name/value pair into its own dialplan line.
-
-* app_read has been updated to use the newer options codes, using "skip" or
-  "noanswer" will not work.  Use s or n.  Also there is a new feature i, for
-  using indication tones, so typing in skip would give you unexpected results.
-
-* OSPAuth is added to authenticate OSP tokens in in_bound call setup messages.
-
-* The CONNECT event in the queue_log from app_queue now has a second field
-  in addition to the holdtime field. It contains the unique ID of the
-  queue member channel that is taking the call. This is useful when trying
-  to link recording filenames back to a particular call from the queue.
-
-* The old/current behavior of app_queue has a serial type behavior
-  in that the queue will make all waiting callers wait in the queue
-  even if there is more than one available member ready to take
-  calls until the head caller is connected with the member they
-  were trying to get to. The next waiting caller in line then
-  becomes the head caller, and they are then connected with the
-  next available member and all available members and waiting callers
-  waits while this happens. This cycle continues until there are
-  no more available members or waiting callers, whichever comes first.
-  The new behavior, enabled by setting autofill=yes in queues.conf
-  either at the [general] level to default for all queues or
-  to set on a per-queue level, makes sure that when the waiting
-  callers are connecting with available members in a parallel fashion
-  until there are no more available members or no more waiting callers,
-  whichever comes first. This is probably more along the lines of how
-  one would expect a queue should work and in most cases, you will want
-  to enable this new behavior. If you do not specify or comment out this
-  option, it will default to "no" to keep backward compatability with the old
-  behavior.
-
-* Queues depend on the channel driver reporting the proper state
-  for each member of the queue. To get proper signalling on
-  queue members that use the SIP channel driver, you need to
-  enable a call limit (could be set to a high value so it
-  is not put into action) and also make sure that both inbound
-  and outbound calls are accounted for.
-
-  Example:
-
-       [general]
-       limitonpeer = yes
-
-       [peername]
-       type=friend
-       call-limit=10
-
-
-* The app_queue application now has the ability to use MixMonitor to
-  record conversations queue members are having with queue callers. Please
-  see configs/queues.conf.sample for more information on this option.
-
-* The app_queue application strategy called 'roundrobin' has been deprecated
-  for this release. Users are encouraged to use 'rrmemory' instead, since it
-  provides more 'true' round-robin call delivery. For the Asterisk 1.6 release,
-  'rrmemory' will be renamed 'roundrobin'.
-
-* The app_queue application option called 'monitor-join' has been deprecated
-  for this release. Users are encouraged to use 'monitor-type=mixmonitor' instead,
-  since it provides the same functionality but is not dependent on soxmix or some
-  other external program in order to mix the audio.
-
-* app_meetme: The 'm' option (monitor) is renamed to 'l' (listen only), and
-  the 'm' option now provides the functionality of "initially muted".
-  In practice, most existing dialplans using the 'm' flag should not notice
-  any difference, unless the keypad menu is enabled, allowing the user
-  to unmute themsleves.
-
-* ast_play_and_record would attempt to cancel the recording if a DTMF
-  '0' was received.  This behavior was not documented in most of the
-  applications that used ast_play_and_record and the return codes from
-  ast_play_and_record weren't checked for properly.
-  ast_play_and_record has been changed so that '0' no longer cancels a
-  recording.  If you want to allow DTMF digits to cancel an
-  in-progress recording use ast_play_and_record_full which allows you
-  to specify which DTMF digits can be used to accept a recording and
-  which digits can be used to cancel a recording.
-
-* ast_app_messagecount has been renamed to ast_app_inboxcount.  There is now a
-  new ast_app_messagecount function which takes a single context/mailbox/folder
-  mailbox specification and returns the message count for that folder only.
-  This addresses the deficiency of not being able to count the number of
-  messages in folders other than INBOX and Old.
-
-* The exit behavior of the AGI applications has changed. Previously, when
-  a connection to an AGI server failed, the application would cause the channel
-  to immediately stop dialplan execution and hangup. Now, the only time that
-  the AGI applications will cause the channel to stop dialplan execution is
-  when the channel itself requests hangup. The AGI applications now set an
-  AGISTATUS variable which will allow you to find out whether running the AGI
-  was successful or not.
-
-  Previously, there was no way to handle the case where Asterisk was unable to
-  locally execute an AGI script for some reason. In this case, dialplan
-  execution will continue as it did before, but the AGISTATUS variable will be
-  set to "FAILURE".
-
-  A locally executed AGI script can now exit with a non-zero exit code and this
-  failure will be detected by Asterisk. If an AGI script exits with a non-zero
-  exit code, the AGISTATUS variable will be set to "FAILURE" as opposed to
-  "SUCCESS".
-
-* app_voicemail: The ODBC_STORAGE capability now requires the extended table format
-  previously used only by EXTENDED_ODBC_STORAGE. This means that you will need to update
-  your table format using the schema provided in doc/odbcstorage.txt
-
-* app_waitforsilence: Fixes have been made to this application which changes the
-  default behavior with how quickly it returns. You can maintain "old-style" behavior
-  with the addition/use of a third "timeout" parameter.
-  Please consult the application documentation and make changes to your dialplan
-  if appropriate.
-
-Manager:
-
-* After executing the 'status' manager action, the "Status" manager events
-  included the header "CallerID:" which was actually only the CallerID number,
-  and not the full CallerID string.  This header has been renamed to
-  "CallerIDNum".  For compatibility purposes, the CallerID parameter will remain
-  until after the release of 1.4, when it will be removed.  Please use the time
-  during the 1.4 release to make this transition.
-
-* The AgentConnect event now has an additional field called "BridgedChannel"
-  which contains the unique ID of the queue member channel that is taking the
-  call. This is useful when trying to link recording filenames back to
-  a particular call from the queue.
-
-* app_userevent has been modified to always send Event: UserEvent with the
-  additional header UserEvent: <userspec>.  Also, the Channel and UniqueID
-  headers are not automatically sent, unless you specify them as separate
-  arguments.  Please see the application help for the new syntax.
-
-* app_meetme: Mute and Unmute events are now reported via the Manager API.
-  Native Manager API commands MeetMeMute and MeetMeUnmute are provided, which
-  are easier to use than "Action Command:". The MeetMeStopTalking event has
-  also been deprecated in favor of the already existing MeetmeTalking event
-  with a "Status" of "on" or "off" added.
-
-* OriginateFailure and OriginateSuccess events were replaced by event
-  OriginateResponse with a header named "Response" to indicate success or
-  failure
-
-Variables:
-
-* The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
-  ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE},
-  and ${LANGUAGE} have all been deprecated in favor of their related dialplan
-  functions.  You are encouraged to move towards the associated dialplan
-  function, as these variables will be removed in a future release.
-
-* The CDR-CSV variables uniqueid, userfield, and basing time on GMT are now
-  adjustable from cdr.conf, instead of recompiling.
-
-* OSP applications exports several new variables, ${OSPINHANDLE},
-  ${OSPOUTHANDLE}, ${OSPINTOKEN}, ${OSPOUTTOKEN}, ${OSPCALLING},
-  ${OSPINTIMELIMIT}, and ${OSPOUTTIMELIMIT}
-
-* Builtin transfer functionality sets the variable ${TRANSFERERNAME} in the new
-  created channel. This variables holds the channel name of the transferer.
-
-* The dial plan variable PRI_CAUSE will be removed from future versions
-  of Asterisk.
-  It is replaced by adding a cause value to the hangup() application.
-
-Functions:
-
-* The function ${CHECK_MD5()} has been deprecated in favor of using an
-  expression: $[${MD5(<string>)} = ${saved_md5}].
-
-* The 'builtin' functions that used to be combined in pbx_functions.so are
-  now built as separate modules. If you are not using 'autoload=yes' in your
-  modules.conf file then you will need to explicitly load the modules that
-  contain the functions you want to use.
-
-* The ENUMLOOKUP() function with the 'c' option (for counting the number of
-  records), but the lookup fails to match any records, the returned value will
-  now be "0" instead of blank.
-
-* The REALTIME() function is now available in version 1.4 and app_realtime has
-  been deprecated in favor of the new function. app_realtime will be removed
-  completely with the version 1.6 release so please take the time between
-  releases to make any necessary changes
-
-* The QUEUEAGENTCOUNT() function has been deprecated in favor of
-  QUEUE_MEMBER_COUNT().
-
-The IAX2 channel:
-
-* It is possible that previous configurations depended on the order in which
-  peers and users were specified in iax.conf for forcing the order in which
-  chan_iax2 matched against them.  This behavior is going away and is considered
-  deprecated in this version.  Avoid having ambiguous peer and user entries and
-  to make things easy on yourself, always set the "username" option for users
-  so that the remote end can match on that exactly instead of trying to infer
-  which user you want based on host.
-
-  If you would like to go ahead and use the new behavior which doesn't use the
-  order in the config file to influence matching order, then change the
-  MAX_PEER_BUCKETS define in chan_iax2.c to a value greater than one.  An
-  example is provided there.  By changing this, you will get *much* better
-  performance on systems that do a lot of peer and user lookups as they will be
-  stored in memory in a much more efficient manner.
-
-* The "mailboxdetail" option has been deprecated.  Previously, if this option
-  was not enabled, the 2 byte MSGCOUNT information element would be set to all
-  1's to indicate there there is some number of messages waiting.  With this
-  option enabled, the number of new messages were placed in one byte and the
-  number of old messages are placed in the other.  This is now the default
-  (and the only) behavior.
-
-The SIP channel:
-
-* The "incominglimit" setting is replaced by the "call-limit" setting in
-  sip.conf.
-
-* OSP support code is removed from SIP channel to OSP applications. ospauth
-  option in sip.conf is removed to osp.conf as authpolicy. allowguest option
-  in sip.conf cannot be set as osp anymore.
-
-* The Asterisk RTP stack has been changed in regards to RFC2833 reception
-  and transmission. Packets will now be sent with proper duration instead of all
-  at once. If you are receiving calls from a pre-1.4 Asterisk installation you
-  will want to turn on the rfc2833compensate option. Without this option your
-  DTMF reception may act poorly.
-
-* The $SIPUSERAGENT dialplan variable is deprecated and will be removed
-  in coming versions of Asterisk. Please use the dialplan function
-  SIPCHANINFO(useragent) instead.
-
-* The ALERT_INFO dialplan variable is deprecated and will be removed
-  in coming versions of Asterisk. Please use the dialplan application
-  sipaddheader() to add the "Alert-Info" header to the outbound invite.
-
-* The "canreinvite" option has changed. canreinvite=yes used to disable
-  re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat
-  to disable re-invites when NAT=yes. This is propably what you want.
-  The settings are now: "yes", "no", "nonat", "update". Please consult
-  sip.conf.sample for detailed information.
-
-The Zap channel:
-
-* Support for MFC/R2 has been removed, as it has not been functional for some
-  time and it has no maintainer.
-
-The Agent channel:
-
-* Callback mode (AgentCallbackLogin) is now deprecated, since the entire function
-  it provided can be done using dialplan logic, without requiring additional
-  channel and module locks (which frequently caused deadlocks). An example of
-  how to do this using AEL dialplan is in doc/queues-with-callback-members.txt.
-
-The G726-32 codec:
-
-* It has been determined that previous versions of Asterisk used the wrong codeword
-  packing order for G726-32 data. This version supports both available packing orders,
-  and can transcode between them. It also now selects the proper order when
-  negotiating with a SIP peer based on the codec name supplied in the SDP. However,
-  there are existing devices that improperly request one order and then use another;
-  Sipura and Grandstream ATAs are known to do this, and there may be others. To
-  be able to continue to use these devices with this version of Asterisk and the
-  G726-32 codec, a configuration parameter called 'g726nonstandard' has been added
-  to sip.conf, so that Asterisk can use the packing order expected by the device (even
-  though it requested a different order). In addition, the internal format number for
-  G726-32 has been changed, and the old number is now assigned to AAL2-G726-32. The
-  result of this is that this version of Asterisk will be able to interoperate over
-  IAX2 with older versions of Asterisk, as long as this version is told to allow
-  'g726aal2' instead of 'g726' as the codec for the call.
-
-Installation:
-
-* On BSD systems, the installation directories have changed to more "FreeBSDish"
-  directories. On startup, Asterisk will look for the main configuration in
-  /usr/local/etc/asterisk/asterisk.conf
-  If you have an old installation, you might want to remove the binaries and
-  move the configuration files to the new locations. The following directories
-  are now default:
-	ASTLIBDIR	/usr/local/lib/asterisk
-	ASTVARLIBDIR	/usr/local/share/asterisk
-	ASTETCDIR	/usr/local/etc/asterisk
-	ASTBINDIR	/usr/local/bin/asterisk
-	ASTSBINDIR	/usr/local/sbin/asterisk
-
-Music on Hold:
-
-* The music on hold handling has been changed in some significant ways in hopes
-  to make it work in a way that is much less confusing to users. Behavior will
-  not change if the same configuration is used from older versions of Asterisk.
-  However, there are some new configuration options that will make things work
-  in a way that makes more sense.
-
-  Previously, many of the channel drivers had an option called "musicclass" or
-  something similar. This option set what music on hold class this channel
-  would *hear* when put on hold. Some people expected (with good reason) that
-  this option was to configure what music on hold class to play when putting
-  the bridged channel on hold. This option has now been deprecated.
-
-  Two new music on hold related configuration options for channel drivers have
-  been introduced. Some channel drivers support both options, some just one,
-  and some support neither of them. Check the sample configuration files to see
-  which options apply to which channel driver.
-
-  The "mohsuggest" option specifies which music on hold class to suggest to the
-  bridged channel when putting them on hold. The only way that this class can
-  be overridden is if the bridged channel has a specific music class set that
-  was done in the dialplan using Set(CHANNEL(musicclass)=something).
-
-  The "mohinterpret" option is similar to the old "musicclass" option. It
-  specifies which music on hold class this channel would like to listen to when
-  put on hold. This music class is only effective if this channel has no music
-  class set on it from the dialplan and the bridged channel putting this one on
-  hold had no "mohsuggest" setting.
-
-  The IAX2 and Zap channel drivers have an additional feature for the
-  "mohinterpret" option. If this option is set to "passthrough", then these
-  channel drivers will pass through the HOLD message in signalling instead of
-  starting music on hold on the channel. An example for how this would be
-  useful is in an enterprise network of Asterisk servers. When one phone on one
-  server puts a phone on a different server on hold, the remote server will be
-  responsible for playing the hold music to its local phone that was put on
-  hold instead of the far end server across the network playing the music.
-
-CDR Records:
-
-* The behavior of the "clid" field of the CDR has always been that it will
-  contain the callerid ANI if it is set, or the callerid number if ANI was not
-  set.  When using the "callerid" option for various channel drivers, some
-  would set ANI and some would not.  This has been cleared up so that all
-  channel drivers set ANI.  If you would like to change the callerid number
-  on the channel from the dialplan and have that change also show up in the
-  CDR, then you *must* set CALLERID(ANI) as well as CALLERID(num).
-
-API:
-
-* There are some API functions that were not previously prefixed with the 'ast_'
-  prefix but now are; these include the ADSI, ODBC and AGI interfaces. If you
-  have a module that uses the services provided by res_adsi, res_odbc, or
-  res_agi, you will need to add ast_ prefixes to the functions that you call
-  from those modules.
-
-Formats:
-
-* format_wav: The GAIN preprocessor definition has been changed from 2 to 0
-  in Asterisk 1.4.  This change was made in response to user complaints of
-  choppiness or the clipping of loud signal peaks.  The GAIN preprocessor
-  definition will be retained in Asterisk 1.4, but will be removed in a
-  future release.  The use of GAIN for the increasing of voicemail message
-  volume should use the 'volgain' option in voicemail.conf

UPGRADE-1.6.txt

@@ -1,277 +0,0 @@
-=========================================================
-===
-=== Information for upgrading from Asterisk 1.4 to 1.6
-===
-=== These files document all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also includes advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
-=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
-===
-=========================================================
-
-AEL:
-
-* Macros are now implemented underneath with the Gosub() application.
-  Heaven Help You if you wrote code depending on any aspect of this!
-  Previous to 1.6, macros were implemented with the Macro() app, which
-  provided a nice feature of auto-returning. The compiler will do its
-  best to insert a Return() app call at the end of your macro if you did
-  not include it, but really, you should make sure that all execution
-  paths within your macros end in "return;".
-
-* The conf2ael program is 'introduced' in this release; it is in a rather
-  crude state, but deemed useful for making a first pass at converting
-  extensions.conf code into AEL. More intelligence will come with time.
-
-Core:
-
-* The 'languageprefix' option in asterisk.conf is now deprecated, and
-  the default sound file layout for non-English sounds is the 'new
-  style' layout introduced in Asterisk 1.4 (and used by the automatic
-  sound file installer in the Makefile).
-
-* The ast_expr2 stuff has been modified to handle floating-point numbers.
-  Numbers of the format D.D are now acceptable input for the expr parser,
-  Where D is a string of base-10 digits. All math is now done in "long double",
-  if it is available on your compiler/architecture. This was half-way between
-  a bug-fix (because the MATH func returns fp by default), and an enhancement.
-  Also, for those counting on, or needing, integer operations, a series of
-  'functions' were also added to the expr language, to allow several styles
-  of rounding/truncation, along with a set of common floating point operations,
-  like sin, cos, tan, log, pow, etc. The ability to call external functions
-  like CDR(), etc. was also added, without having to use the ${...} notation.
-
-* The delimiter passed to applications has been changed to the comma (','), as
-  that is what people are used to using within extensions.conf.  If you are
-  using realtime extensions, you will need to translate your existing dialplan
-  to use this separator.  To use a literal comma, you need merely to escape it
-  with a backslash ('\').  Another possible side effect is that you may need to
-  remove the obscene level of backslashing that was necessary for the dialplan
-  to work correctly in 1.4 and previous versions.  This should make writing
-  dialplans less painful in the future, albeit with the pain of a one-time
-  conversion.  If you would like to avoid this conversion immediately, set
-  pbx_realtime=1.4 in the [compat] section of asterisk.conf.  After
-  transitioning, set pbx_realtime=1.6 in the same section.
-
-* For the same purpose as above, you may set res_agi=1.4 in the [compat]
-  section of asterisk.conf to continue to use the '|' delimiter in the EXEC
-  arguments of AGI applications.  After converting to use the ',' delimiter,
-  change this option to res_agi=1.6.
-
-* As a side effect of the application delimiter change, many places that used
-  to need quotes in order to get the proper meaning are no longer required.
-  You now only need to quote strings in configuration files if you literally
-  want quotation marks within a string.
-
-* Any applications run that contain the pipe symbol but not a comma symbol will
-  get a warning printed to the effect that the application delimiter has changed.
-  However, there are legitimate reasons why this might be useful in certain
-  situations, so this warning can be turned off with the dontwarn option in
-  asterisk.conf.
-
-* The logger.conf option 'rotatetimestamp' has been deprecated in favor of
-  'rotatestrategy'.  This new option supports a 'rotate' strategy that more
-  closely mimics the system logger in terms of file rotation.
-
-* The concise versions of various CLI commands are now deprecated. We recommend
-  using the manager interface (AMI) for application integration with Asterisk.
-
-Voicemail:
-
-* The voicemail configuration values 'maxmessage' and 'minmessage' have
-  been changed to 'maxsecs' and 'minsecs' to clarify their purpose and
-  to make them more distinguishable from 'maxmsgs', which sets folder
-  size.  The old variables will continue to work in this version, albeit
-  with a deprecation warning.
-
-* If you use any interface for modifying voicemail aside from the built in
-  dialplan applications, then the option "pollmailboxes" *must* be set in
-  voicemail.conf for message waiting indication (MWI) to work properly.  This
-  is because Voicemail notification is now event based instead of polling
-  based.  The channel drivers are no longer responsible for constantly manually
-  checking mailboxes for changes so that they can send MWI information to users.
-  Examples of situations that would require this option are web interfaces to
-  voicemail or an email client in the case of using IMAP storage.
-
-Applications:
-
-
-* ChanIsAvail() now has a 't' option, which allows the specified device
-  to be queried for state without consulting the channel drivers. This
-  performs mostly a 'ChanExists' sort of function.
-
-* ChannelRedirect() will not terminate the channel that fails to do a
-  channelredirect as it has done previously. Instead CHANNELREDIRECT_STATUS
-  will reflect if the attempt was successful of not.
-
-* SetCallerPres() has been replaced with the CALLERPRES() dialplan function
-  and is now deprecated.
-
-* DISA()'s fifth argument is now an options argument.  If you have previously
-  used 'NOANSWER' in this argument, you'll need to convert that to the new
-  option 'n'.
-
-* Macro() is now deprecated.  If you need subroutines, you should use the
-  Gosub()/Return() applications.  To replace MacroExclusive(), we have
-  introduced dialplan functions LOCK(), TRYLOCK(), and UNLOCK().  You may use
-  these functions in any location where you desire to ensure that only one
-  channel is executing that path at any one time.  The Macro() applications
-  are deprecated for performance reasons.  However, since Macro() has been
-  around for a long time and so many dialplans depend heavily on it, for the
-  sake of backwards compatibility it will not be removed .  It is also worth
-  noting that using both Macro() and GoSub() at the same time is _heavily_
-  discouraged.
-
-* Read() now sets a READSTATUS variable on exit.  It does NOT automatically
-  return -1 (and hangup) anymore on error.  If you want to hangup on error,
-  you need to do so explicitly in your dialplan.
-
-* Privacy() no longer uses privacy.conf, so any options must be specified
-  directly in the application arguments.
-
-* MusicOnHold application now has duration parameter which allows specifying
-  timeout in seconds.
-
-* WaitMusicOnHold application is now deprecated in favor of extended MusicOnHold.
-
-* SetMusicOnHold is now deprecated. You should use Set(CHANNEL(musicclass)=...)
-  instead.
-
-* The arguments in ExecIf changed a bit, to be more like other applications.
-  The syntax is now ExecIf(<cond>?appiftrue(args):appiffalse(args)).
-
-* The behavior of the Set application now depends upon a compatibility option,
-  set in asterisk.conf.  To use the old 1.4 behavior, which allowed Set to take
-  multiple key/value pairs, set app_set=1.4 in [compat] in asterisk.conf.  To
-  use the new behavior, which permits variables to be set with embedded commas,
-  set app_set=1.6 in [compat] in asterisk.conf.  Note that you can have both
-  behaviors at the same time, if you switch to using MSet if you want the old
-  behavior.
-
-Dialplan Functions:
-
-* QUEUE_MEMBER_COUNT() has been deprecated in favor of the QUEUE_MEMBER() function. For
-  more information, issue a "show function QUEUE_MEMBER" from the CLI.
-
-CDR:
-
-* The cdr_sqlite module has been marked as deprecated in favor of
-  cdr_sqlite3_custom.  It will potentially be removed from the tree
-  after Asterisk 1.6 is released.
-
-* The cdr_odbc module now uses res_odbc to manage its connections.  The
-  username and password parameters in cdr_odbc.conf, therefore, are no
-  longer used.  The dsn parameter now points to an entry in res_odbc.conf.
-
-* The uniqueid field in the core Asterisk structure has been changed from a
-  maximum 31 character field to a 149 character field, to account for all
-  possible values the systemname prefix could be.  In the past, if the
-  systemname was too long, the uniqueid would have been truncated.
-
-* The cdr_tds module now supports all versions of FreeTDS that contain
-  the db-lib frontend.  It will also now log the userfield variable if
-  the target database table contains a column for it.
-
-Formats:
-
-* format_wav: The GAIN preprocessor definition and source code that used it
-  is removed.  This change was made in response to user complaints of
-  choppiness or the clipping of loud signal peaks.  To increase the volume
-  of voicemail messages, use the 'volgain' option in voicemail.conf
-
-Channel Drivers:
-
-* SIP: a small upgrade to support the "Record" button on the SNOM360,
-  which sends a sip INFO message with a "Record: on" or "Record: off"
-  header. If Asterisk is set up (via features.conf) to accept "One Touch Monitor"
-  requests (by default, via '*1'), then the user-configured dialpad sequence
-  is generated, and recording can be started and stopped via this button. The
-  file names and formats are all controlled via the normal mechanisms. If the
-  user has not configured the automon feature, the normal "415 Unsupported media type"
-  is returned, and nothing is done.
-
-* SIP: The "call-limit" option is marked as deprecated. It still works in this version of
-  Asterisk, but will be removed in the following version. Please use the groupcount functions
-  in the dialplan to enforce call limits. The "limitonpeer" configuration option is
-  now renamed to "counteronpeer".
-
-* SIP: The "username" option is now renamed to "defaultuser" to match "defaultip".
-  These are used only before registration to call a peer with the uri
-	sip:defaultuser@defaultip
-  The "username" setting still work, but is deprecated and will not work in
-  the next version of Asterisk.
-
-* SIP: The old "insecure" options, deprecated in 1.4, have been removed.
-  "insecure=very" should be changed to "insecure=port,invite"
-  "insecure=yes" should be changed to "insecure=port"
-  Be aware that some telephony providers show the invalid syntax in their
-  sample configurations.
-
-* chan_local.c: the comma delimiter inside the channel name has been changed to a
-  semicolon, in order to make the Local channel driver compatible with the comma
-  delimiter change in applications.
-
-* H323: The "tos" setting has changed name to "tos_audio" and "cos" to "cos_audio"
-  to be compatible with settings in sip.conf. The "tos" and "cos" configuration
-  is deprecated and will stop working in the next release of Asterisk.
-
-* Console: A new console channel driver, chan_console, has been added to Asterisk.
-  This new module can not be loaded at the same time as chan_alsa or chan_oss.  The
-  default modules.conf only loads one of them (chan_oss by default).  So, unless you
-  have modified your modules.conf to not use the autoload option, then you will need
-  to modify modules.conf to add another "noload" line to ensure that only one of
-  these three modules gets loaded.
-
-* DAHDI: The chan_zap module that supported PSTN interfaces using
-  Zaptel has been renamed to chan_dahdi, and only supports the DAHDI
-  telephony driver package for PSTN interfaces. See the
-  Zaptel-to-DAHDI.txt file for more details on this transition.
-
-* DAHDI: The "msdstrip" option has been deprecated, as it provides no value over
-  the method of stripping digits in the dialplan using variable substring syntax.
-
-Configuration:
-
-* pbx_dundi.c: tos parameter changed to use new values. Old values like lowdelay,
-  lowcost and other is not acceptable now. Look into qos.tex for description of
-  this parameter.
-
-* queues.conf: the queue-lessthan sound file option is no longer available, and the
-  queue-round-seconds option no longer takes '1' as a valid parameter.
-
-Manager:
-
-* Manager has been upgraded to version 1.1 with a lot of changes.
-  Please check doc/manager_1_1.txt for information
-
-* The IAXpeers command output has been changed to more closely resemble the
-  output of the SIPpeers command.
-
-* cdr_manager now reports at the "cdr" level, not at "call"  You may need to
-   change your manager.conf to add the level to existing AMI users, if they
-   want to see the CDR events generated.
-
-* The Originate command now requires the Originate write permission.  For
-   Originate with the Application parameter, you need the additional System
-   privilege if you want to do anything that calls out to a subshell.
-
-iLBC Codec:
-
-* Previously, the Asterisk source code distribution included the iLBC
-  encoder/decoder source code, from Global IP Solutions
-  (http://www.gipscorp.com). This code is not licensed for
-  distribution, and thus has been removed from the Asterisk source
-  code distribution. If you wish to use codec_ilbc to support iLBC
-  channels in Asterisk, you can run the contrib/scripts/get_ilbc_source.sh
-  script to download the source and put it in the proper place in
-  the Asterisk build tree. Once that is done you can follow your normal
-  steps of building Asterisk. You will need to run 'menuselect' and enable
-  the iLBC codec in the 'Codec  Translators' category.

UPGRADE-1.8.txt

@@ -1,343 +0,0 @@
-===========================================================
-===
-=== Information for upgrading between Asterisk versions
-===
-=== These files document all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also includes advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
-=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
-=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
-===
-===========================================================
-
-From 1.8.13 to 1.8.14:
-* permitdirectmedia/denydirectmedia now controls whether peers can be
-  bridged via directmedia by comparing the ACL to the bridging peer's
-  address rather than its own address.
-
-From 1.8.12 to 1.8.13:
-* The complex processor detection and optimization has been removed from
-  the makefile in favor of using native optimization suppport when available.
-  BUILD_NATIVE can be disabled via menuselect under "Compiler Flags".
-
-From 1.8.10 to 1.8.11:
-
-* If no transport is specified in sip.conf, transport will default to UDP.
-  Also, if multiple transport= lines are used, only the last will be used.
-
-From 1.6.2 to 1.8:
-
-* chan_sip no longer sets HASH(SIP_CAUSE,<chan name>) on channels by default.
-  This must now be enabled by setting 'sipstorecause' to 'yes' in sip.conf.
-  This carries a performance penalty.
-
-* Asterisk now requires libpri 1.4.11+ for PRI support.
-
-* A couple of CLI commands in res_ais were changed back to their original form:
-    "ais show clm members" --> "ais clm show members"
-    "ais show evt event channels" --> "ais evt show event channels"
-
-* The default value for 'autofill' and 'shared_lastcall' in queues.conf has
-  been changed to 'yes'.
-
-* The default value for the alwaysauthreject option in sip.conf has been changed
-  from "no" to "yes".
-
-* The behavior of the 'parkedcallstimeout' has changed slightly.  The formulation
-  of the extension name that a timed out parked call is delivered to when this
-  option is set to 'no' was modified such that instead of converting '/' to '0',
-  the '/' is converted to an underscore '_'.  See the updated documentation in
-  features.conf.sample for more information on the behavior of the
-  'parkedcallstimeout' option.
-
-* Asterisk-addons no longer exists as an independent package.  Those modules
-  now live in the addons directory of the main Asterisk source tree.  They
-  are not enabled by default.  For more information about why modules live in
-  addons, see README-addons.txt.
-
-* The rarely used 'event_log' and LOG_EVENT channel have been removed; the few
-  users of this channel in the tree have been converted to LOG_NOTICE or removed
-  (in cases where the same message was already generated to another channel).
-
-* The usage of RTP inside of Asterisk has now become modularized. This means
-  the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.
-  If you are not using autoload=yes in modules.conf you will need to ensure
-  it is set to load. If not, then any module which uses RTP (such as chan_sip)
-  will not be able to send or receive calls.
-
-* The app_dahdiscan.c file has been removed, but the dialplan app DAHDIScan still
-  remains. It now exists within app_chanspy.c and retains the exact same
-  functionality as before.
-
-* The default behavior for Set, AGI, and pbx_realtime has been changed to implement
-  1.6 behavior by default, if there is no [compat] section in asterisk.conf.  In
-  prior versions, the behavior defaulted to 1.4 behavior, to assist in upgrades.
-  Specifically, that means that pbx_realtime and res_agi expect you to use commas
-  to separate arguments in applications, and Set only takes a single pair of
-  a variable name/value.  The old 1.4 behavior may still be obtained by setting
-  app_set, pbx_realtime, and res_agi each to 1.4 in the [compat] section of
-  asterisk.conf.
-
-* The PRI channels in chan_dahdi can no longer change the channel name if a
-  different B channel is selected during call negotiation.  To prevent using
-  the channel name to infer what B channel a call is using and to avoid name
-  collisions, the channel name format is changed.
-  The new channel naming for PRI channels is:
-  DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
-
-* Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type)
-  so the dialplan can determine the B channel currently in use by the channel.
-  Use CHANNEL(no_media_path) to determine if the channel even has a B channel.
-
-* Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk
-  channel so AMI applications can passively determine the B channel currently
-  in use.  Calls with "no-media" as the DAHDIChannel do not have an associated
-  B channel.  No-media calls are either on hold or call-waiting.
-
-* The ChanIsAvail application has been changed so the AVAILSTATUS variable
-  no longer contains both the device state and cause code. The cause code
-  is now available in the AVAILCAUSECODE variable. If existing dialplan logic
-  is written to expect AVAILSTATUS to contain the cause code it needs to be
-  changed to use AVAILCAUSECODE.
-
-* ExternalIVR will now send Z events for invalid or missing files, T events
-  now include the interrupted file and bugs in argument parsing have been
-  fixed so there may be arguments specified in incorrect ways that were
-  working that will no longer work. Please see
-  https://wiki.asterisk.org/wiki/display/AST/External+IVR+Interface for details.
-
-* OSP lookup application changes following variable names:
-  OSPPEERIP to OSPINPEERIP
-  OSPTECH to OSPOUTTECH
-  OSPDEST to OSPDESTINATION
-  OSPCALLING to OSPOUTCALLING
-  OSPCALLED to OSPOUTCALLED
-  OSPRESULTS to OSPDESTREMAILS
-
-* The Manager event 'iax2 show peers' output has been updated.  It now has a
-  similar output of 'sip show peers'.
-
-* VoiceMailMain and VMAuthenticate, if a '*' is entered in the first position
-  of a Mailbox or Password, will, if it exists, jump to the 'a' extension in
-  the current dialplan context.
-
-* The CALLERPRES() dialplan function is deprecated in favor of
-  CALLERID(num-pres) and CALLERID(name-pres).
-
-* Environment variables that start with "AST_" are reserved to the system and
-  may no longer be set from the dialplan.
-
-* When a call is redirected inside of a Dial, the app and appdata fields of the
-  CDR will now be set to "AppDial" and "(Outgoing Line)" instead of being blank.
-
-* The CDR handling of billsec and duration field has changed. If your table
-  definition specifies those fields as float,double or similar they will now
-  be logged with microsecond accuracy instead of a whole integer.
-
-* chan_sip will no longer set up a local call forward when receiving a
-  482 Loop Detected response. The dialplan will just continue from where it
-  left off.
-
-* The 'stunaddr' option has been removed from chan_sip.  This feature did not
-  behave as expected, had no correct use case, and was not RFC compliant. The
-  removal of this feature will hopefully be followed by a correct RFC compliant
-  STUN implementation in chan_sip in the future.
-
-* The default value for the pedantic option in sip.conf has been changed
-  from "no" to "yes".
-
-* The ConnectedLineNum and ConnectedLineName headers were added to many AMI
-  events/responses if the CallerIDNum/CallerIDName headers were also present.
-  The addition of connected line support changes the behavior of the channel
-  caller ID somewhat.  The channel caller ID value no longer time shares with
-  the connected line ID on outgoing call legs.  The timing of some AMI
-  events/responses output the connected line ID as caller ID.  These party ID's
-  are now separate.
-
-* The Dial application d and H options do not automatically answer the call
-  anymore.  It broke DTMF attended transfers.  Since many SIP and ISDN phones
-  cannot send DTMF before a call is connected, you need to answer the call
-  leg to those phones before using Dial with these options for them to have
-  any effect before the dialed party answers.
-
-* The outgoing directory (where .call files are read) now uses inotify to
-  detect file changes instead of polling the directory on a regular basis.
-  If your outgoing folder is on a NFS mount or another network file system,
-  changes to the files will not be detected.  You can revert to polling the
-  directory by specifying --without-inotify to configure before compiling.
-
-* The 'sipusers' realtime table has been removed completely. Use the 'sippeers'
-  table with type 'user' for user type objects.
-
-* The sip.conf allowoverlap option now accepts 'dtmf' as a value.  If you
-  are using the early media DTMF overlap dialing method you now need to set
-  allowoverlap=dtmf.
-
-From 1.6.1 to 1.6.2:
-
-* SIP no longer sends the 183 progress message for early media by
-  default.  Applications requiring early media should use the
-  progress() dialplan app to generate the progress message.
-
-* The firmware for the IAXy has been removed from Asterisk.  It can be
-  downloaded from http://downloads.digium.com/pub/iaxy/.  To have Asterisk
-  install the firmware into its proper location, place the firmware in the
-  contrib/firmware/iax/ directory in the Asterisk source tree before running
-  "make install".
-
-* T.38 FAX error correction mode can no longer be configured in udptl.conf;
-  instead, it is configured on a per-peer (or global) basis in sip.conf, with
-  the same default as was present in udptl.conf.sample.
-
-* T.38 FAX maximum datagram size can no longer be configured in updtl.conf;
-  instead, it is either supplied by the application servicing the T.38 channel
-  (for a FAX send or receive) or calculated from the bridged endpoint's
-  maximum datagram size (for a T.38 FAX passthrough call). In addition, sip.conf
-  allows for overriding the value supplied by a remote endpoint, which is useful
-  when T.38 connections are made to gateways that supply incorrectly-calculated
-  maximum datagram sizes.
-
-* There have been some changes to the IAX2 protocol to address the security
-  concerns documented in the security advisory AST-2009-006.  Please see the
-  IAX2 security document, doc/IAX2-security.pdf, for information regarding
-  backwards compatibility with versions of Asterisk that do not contain these
-  changes to IAX2.
-
-* The 'canreinvite' option support by the SIP, MGCP and Skinny channel drivers
-  has been renamed to 'directmedia', to better reflect what it actually does.
-  In the case of SIP, there are still re-INVITEs issued for T.38 negotiation,
-  starting and stopping music-on-hold, and other reasons, and the 'canreinvite'
-  option never had any effect on these cases, it only affected the re-INVITEs
-  used for direct media path setup. For MGCP and Skinny, the option was poorly
-  named because those protocols don't even use INVITE messages at all. For
-  backwards compatibility, the old option is still supported in both normal
-  and Realtime configuration files, but all of the sample configuration files,
-  Realtime/LDAP schemas, and other documentation refer to it using the new name.
-
-* The default console now will use colors according to the default background
-  color, instead of forcing the background color to black.  If you are using a
-  light colored background for your console, you may wish to use the option
-  flag '-W' to present better color choices for the various messages.  However,
-  if you'd prefer the old method of forcing colors to white text on a black
-  background, the compatibility option -B is provided for this purpose.
-
-* SendImage() no longer hangs up the channel on transmission error or on
-  any other error; in those cases, a FAILURE status is stored in
-  SENDIMAGESTATUS and dialplan execution continues.  The possible
-  return values stored in SENDIMAGESTATUS are: SUCCESS, FAILURE, and
-  UNSUPPORTED. ('OK' has been replaced with 'SUCCESS', and 'NOSUPPORT'
-  has been replaced with 'UNSUPPORTED').  This change makes the
-  SendImage application more consistent with other applications.
-
-* skinny.conf now has separate sections for lines and devices.
-  Please have a look at configs/skinny.conf.sample and update
-  your skinny.conf.
-
-* Queue names previously were treated in a case-sensitive manner,
-  meaning that queues with names like "sales" and "sALeS" would be
-  seen as unique queues. The parsing logic has changed to use
-  case-insensitive comparisons now when originally hashing based on
-  queue names, meaning that now the two queues mentioned as examples
-  earlier will be seen as having the same name.
-
-* The SPRINTF() dialplan function has been moved into its own module,
-  func_sprintf, and is no longer included in func_strings. If you use this
-  function and do not use 'autoload=yes' in modules.conf, you will need
-  to explicitly load func_sprintf for it to be available.
-
-* The res_indications module has been removed.  Its functionality was important
-  enough that most of it has been moved into the Asterisk core.
-  Two applications previously provided by res_indications, PlayTones and
-  StopPlayTones, have been moved into a new module, app_playtones.
-
-* Support for Taiwanese was incorrectly supported with the "tw" language code.
-  In reality, the "tw" language code is reserved for the Twi language, native
-  to Ghana.  If you were previously using the "tw" language code, you should
-  switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan
-  specific localizations.  Additionally, "mx" should be changed to "es_MX",
-  Georgian was incorrectly specified as "ge" but should be "ka", and Czech is
-  "cs", not "cz".
-
-* DAHDISendCallreroutingFacility() parameters are now comma-separated,
-  instead of the old pipe.
-
-* res_jabber: autoprune has been disabled by default, to avoid misconfiguration
-  that would end up being interpreted as a bug once Asterisk started removing
-  the contacts from a user list.
-
-* The cdr.conf file must exist and be configured correctly in order for CDR
-  records to be written.
-
-* cdr_pgsql now assumes the encoding of strings it is handed are in LATIN9,
-  which should cover most uses of the extended ASCII set.  If your strings
-  use a different encoding in Asterisk, the "encoding" parameter may be set
-  to specify the correct character set.
-
-From 1.6.0.1 to 1.6.1:
-
-* The ast_agi_register_multiple() and ast_agi_unregister_multiple()
-  API calls were added in 1.6.0, so that modules that provide multiple
-  AGI commands could register/unregister them all with a single
-  step. However, these API calls were not implemented properly, and did
-  not allow the caller to know whether registration or unregistration
-  succeeded or failed. They have been redefined to now return success
-  or failure, but this means any code using these functions will need
-  be recompiled after upgrading to a version of Asterisk containing
-  these changes. In addition, the source code using these functions
-  should be reviewed to ensure it can properly react to failure
-  of registration or unregistration of its API commands.
-
-* The ast_agi_fdprintf() API call has been renamed to ast_agi_send()
-  to better match what it really does, and the argument order has been
-  changed to be consistent with other API calls that perform similar
-  operations.
-
-From 1.6.0.x to 1.6.1:
-
-* In previous versions of Asterisk, due to the way objects were arranged in
-  memory by chan_sip, the order of entries in sip.conf could be adjusted to
-  control the behavior of matching against peers and users.  The way objects
-  are managed has been significantly changed for reasons involving performance
-  and stability.  A side effect of these changes is that the order of entries
-  in sip.conf can no longer be relied upon to control behavior.
-
-* The following core commands dealing with dialplan have been deprecated: 'core
-  show globals', 'core set global' and 'core set chanvar'. Use the equivalent
-  'dialplan show globals', 'dialplan set global' and 'dialplan set chanvar'
-  instead.
-
-* In the dialplan expression parser, the logical value of spaces
-  immediately preceding a standalone 0 previously evaluated to
-  true. It now evaluates to false.  This has confused a good many
-  people in the past (typically because they failed to realize the
-  space had any significance).  Since this violates the Principle of
-  Least Surprise, it has been changed.
-
-* While app_directory has always relied on having a voicemail.conf or users.conf file
-  correctly set up, it now is dependent on app_voicemail being compiled as well.
-
-* SIP: All of the functionality in SIPCHANINFO() has been implemented in CHANNEL(),
-  and you should start using that function instead for retrieving information about
-  the channel in a technology-agnostic way.
-
-* If you have any third party modules which use a config file variable whose
-  name ends in a '+', please note that the append capability added to this
-  version may now conflict with that variable naming scheme.  An easy
-  workaround is to ensure that a space occurs between the '+' and the '=',
-  to differentiate your variable from the append operator.  This potential
-  conflict is unlikely, but is documented here to be thorough.
-
-* The "Join" event from app_queue now uses the CallerIDNum header instead of
-  the CallerID header to indicate the CallerID number.
-
-* If you use ODBC storage for voicemail, there is a new field called "flag"
-  which should be a char(8) or larger.  This field specifies whether or not a
-  message has been designated to be "Urgent", "PRIORITY", or not.

UPGRADE-10.txt

@@ -1,92 +0,0 @@
-===========================================================
-===
-=== Information for upgrading between Asterisk versions
-===
-=== These files document all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also include advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
-=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
-=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
-=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
-===
-===========================================================
-
-From 10.4 to 10.5:
-
-* The complex processor detection and optimization has been removed from
-  the makefile in favor of using native optimization suppport when available.
-  BUILD_NATIVE can be disabled via menuselect under "Compiler Flags".
-
-From 10.2 to 10.3:
-
-* If no transport is specified in sip.conf, transport will default to UDP.
-  Also, if multiple transport= lines are used, only the last will be used.
-
-From 1.8 to 10:
-
-cel_pgsql:
- - This module now expects an 'extra' column in the database for data added
-   using the CELGenUserEvent() application.
-
-ConfBridge
- - ConfBridge's dialplan arguments have changed and are not
-   backwards compatible.
-
-File Interpreters
- - The format interpreter formats/format_sln16.c for the file extension
-   '.sln16' has been removed. The '.sln16' file interpreter now exists
-   in the formats/format_sln.c module along with new support for sln12,
-   sln24, sln32, sln44, sln48, sln96, and sln192 file extensions.
-
-HTTP:
- - A bindaddr must be specified in order for the HTTP server
-   to run. Previous versions would default to 0.0.0.0 if no
-   bindaddr was specified.
-
-Gtalk:
- - The default value for 'context' and 'parkinglots' in gtalk.conf has
-   been changed to 'default', previously they were empty.
-
-chan_dahdi:
- - The mohinterpret=passthrough setting is deprecated in favor of
-   moh_signaling=notify.
-
-pbx_lua:
- - Execution no longer continues after applications that do dialplan jumps
-   (such as app.goto).  Now when an application such as app.goto() is called,
-   control is returned back to the pbx engine and the current extension
-   function stops executing.
- - the autoservice now defaults to being on by default
- - autoservice_start() and autoservice_start() no longer return a value.
-
-Queue:
- - Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
- - QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.
-
-Asterisk Database:
- - The internal Asterisk database has been switched from Berkeley DB 1.86 to
-   SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
-   utility in the UTILS section of menuselect. If an existing astdb is found and no
-   astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
-   convert an existing astdb to the SQLite3 version automatically at runtime.
-
-Module Support Level
- - All modules in the addons, apps, bridge, cdr, cel, channels, codecs,
-   formats, funcs, pbx, and res have been updated to include MODULEINFO data
-   that includes <support_level> tags with a value of core, extended, or deprecated.
-   More information is available on the Asterisk wiki at
-   https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
-
-   Deprecated modules are now marked to not build by default and must be explicitly
-   enabled in menuselect.
-
-===========================================================
-===========================================================

UPGRADE-11.txt

@@ -1,280 +0,0 @@
-===========================================================
-===
-=== Information for upgrading between Asterisk versions
-===
-=== These files document all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also include advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
-=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
-=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
-=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
-=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
-===
-===========================================================
-
-From 11.6 to 11.7:
-ConfBridge
- - ConfBridge now has the ability to set the language of announcements to the
-   conference.  The language can be set on a bridge profile in confbridge.conf
-   or by the dialplan function CONFBRIDGE(bridge,language)=en.
-chan_sip - Clarify The "sip show peers" Forcerport Column And Add Comedia
- - Under the "Forcerport" column, the "N" used to mean NAT (i.e. Yes).  With
-   the additon of auto_* NAT settings, the meaning changed and there was a
-   certain combination of letters added to indicate the current setting. The
-   combination of using "Y", "N", "A" or "a", can be confusing.  Therefore, we
-   now display clearly what the current Forcerport setting is: "Yes", "No",
-   "Auto (Yes)", "Auto (No)".
- - Since we are clarifying the Forcerport column, we have added a column to
-   display the Comedia setting since this is useful information as well.  We
-   no longer have a simple "NAT" setting like other versions before 11.
-
-From 11.5 to 11.6:
-* res_agi will now properly indicate if there was an error in streaming an
-  audio file.  The result code will be -1 and the result returned from the
-  the function will be RESULT_FAILURE instead of the prior behavior of always
-  returning RESULT_SUCCESS even if there was an error.
-
-From 11.4 to 11.5:
-* The default settings for chan_sip are now overriden properly by the general
-  settings in sip.conf.  Please look over your settings upon upgrading.
-
-From 11.3 to 11.4:
-* Added the 'n' option to MeetMe to prevent application of the DENOISE function
-  to a channel joining a conference. Some channel drivers that vary the number
-  of audio samples in a voice frame will experience significant quality problems
-  if a denoiser is attached to the channel; this option gives them the ability
-  to remove the denoiser without having to unload func_speex.
-
-* The Registry AMI event for SIP registrations will now always include the
-  Username field. A previous bug fix missed an instance where it was not
-  included; that has been corrected in this release.
-
-From 11.2.0 to 11.2.1:
-* Asterisk would previously not output certain error messages when a remote
-  console attempted to connect to Asterisk and no instance of Asterisk was
-  running. This error message is displayed on stderr; as a result, some
-  initialization scripts that used remote consoles to test for the presence
-  of a running Asterisk instance started to display erroneous error messages.
-  The init.d scripts and the safe_asterisk have been updated in the contrib
-  folder to account for this.
-
-From 11.2 to 11.3:
-
-* Now by default, when Asterisk is installed in a path other than /usr, the
-  Asterisk binary will search for shared libraries in ${libdir} in addition to
-  searching system libraries. This allows Asterisk to find its shared
-  libraries without having to specify LD_LIBRARY_PATH. This can be disabled by
-  passing --disable-rpath to configure.
-
-From 10 to 11:
-
-Voicemail:
- - All voicemails now have a "msg_id" which uniquely identifies a message. For
-   users of filesystem and IMAP storage of voicemail, this should be transparent.
-   For users of ODBC, you will need to add a "msg_id" column to your voice mail
-   messages table. This should be a string capable of holding at least 32 characters.
-   All messages created in old Asterisk installations will have a msg_id added to
-   them when required. This operation should be transparent as well.
-
-Parking:
- - The comebacktoorigin setting must now be set per parking lot. The setting in
-   the general section will not be applied automatically to each parking lot.
- - The BLINDTRANSFER channel variable is deleted from a channel when it is
-   bridged to prevent subtle bugs in the parking feature.  The channel
-   variable is used by Asterisk internally for the Park application to work
-   properly.  If you were using it for your own purposes, copy it to your
-   own channel variable before the channel is bridged.
-
-res_ais:
- - Users of res_ais in versions of Asterisk prior to Asterisk 11 must change
-   to use the res_corosync module, instead.  OpenAIS is deprecated, but
-   Corosync is still actively developed and maintained.  Corosync came out of
-   the OpenAIS project.
-
-Dialplan Functions:
- - MAILBOX_EXISTS has been deprecated. Use VM_INFO with the 'exists' parameter
-   instead.
- - Macro has been deprecated in favor of GoSub.  For redirecting and connected
-   line purposes use the following variables instead of their macro equivalents:
-   REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS,
-   CONNECTED_LINE_SEND_SUB, CONNECTED_LINE_SEND_SUB_ARGS.
- - The REDIRECTING function now supports the redirecting original party id
-   and reason.
- - The HANGUPCAUSE and HANGUPCAUSE_KEYS functions have been introduced to
-   provide a replacement for the SIP_CAUSE hash. The HangupCauseClear
-   application has also been introduced to remove this data from the channel
-   when necessary.
-
-
-func_enum:
- - ENUM query functions now return a count of -1 on lookup error to
-   differentiate between a failed query and a successful query with 0 results
-   matching the specified type.
-
-CDR:
- - cdr_adaptive_odbc now supports specifying a schema so that Asterisk can
-   connect to databases that use schemas.
-
-Configuration Files:
- - Files listed below have been updated to be more consistent with how Asterisk
-   parses configuration files.  This makes configuration files more consistent
-   with what is expected across modules.
-
-   - cdr.conf: [general] and [csv] sections
-   - dnsmgr.conf
-   - dsp.conf
-
- - The 'verbose' setting in logger.conf now takes an optional argument,
-   specifying the verbosity level for each logging destination.  The default,
-   if not otherwise specified, is a verbosity of 3.
-
-AMI:
-  - DBDelTree now correctly returns an error when 0 rows are deleted just as
-    the DBDel action does.
-  - The IAX2 PeerStatus event now sends a 'Port' header.  In Asterisk 10, this was
-    erroneously being sent as a 'Post' header.
-
-CCSS:
- - Macro is deprecated. Use cc_callback_sub instead of cc_callback_macro
-   in channel configurations.
-
-app_meetme:
-  - The 'c' option (announce user count) will now work even if the 'q' (quiet)
-    option is enabled.
-
-app_followme:
- - Answered outgoing calls no longer get cut off when the next step is started.
-   You now have until the last step times out to decide if you want to accept
-   the call or not before being disconnected.
-
-chan_gtalk:
- - chan_gtalk has been deprecated in favor of the chan_motif channel driver. It is recommended
-   that users switch to using it as it is a core supported module.
-
-chan_jingle:
- - chan_jingle has been deprecated in favor of the chan_motif channel driver. It is recommended
-   that users switch to using it as it is a core supported module.
-
-SIP
-===
- - A new option "tonezone" for setting default tonezone for the channel driver
-   or individual devices
- - A new manager event, "SessionTimeout" has been added and is triggered when
-   a call is terminated due to RTP stream inactivity or SIP session timer
-   expiration.
- - SIP_CAUSE is now deprecated.  It has been modified to use the same
-   mechanism as the HANGUPCAUSE function.  Behavior should not change, but
-   performance should be vastly improved.  The HANGUPCAUSE function should now
-   be used instead of SIP_CAUSE. Because of this, the storesipcause option in
-   sip.conf is also deprecated.
- - The sip paramater for Originating Line Information (oli, isup-oli, and
-   ss7-oli) is now parsed out of the From header and copied into the channel's
-   ANI2 information field.  This is readable from the CALLERID(ani2) dialplan
-   function.
- - ICE support has been added and is enabled by default. Some endpoints may have
-   problems with the ICE candidates within the SDP. If this is the case ICE support
-   can be disabled globally or on a per-endpoint basis using the icesupport
-   configuration option. Symptoms of this include one way media or no media flow.
-
-chan_unistim
- - Due to massive update in chan_unistim phone keys functions and on-screen
-   information changed.
-
-users.conf:
- - A defined user with hasvoicemail=yes now finally uses a Gosub to stdexten
-   as documented in extensions.conf.sample since v1.6.0 instead of a Macro as
-   documented in v1.4.  Set the asterisk.conf stdexten=macro parameter to
-   invoke the stdexten the old way.
-
-res_jabber
- - This module has been deprecated in favor of the res_xmpp module. The res_xmpp
-   module is backwards compatible with the res_jabber configuration file, dialplan
-   functions, and AMI actions. The old CLI commands can also be made available using
-   the res_clialiases template for Asterisk 11.
-
-From 1.8 to 10:
-
-cel_pgsql:
- - This module now expects an 'extra' column in the database for data added
-   using the CELGenUserEvent() application.
-
-ConfBridge
- - ConfBridge's dialplan arguments have changed and are not
-   backwards compatible.
-
-File Interpreters
- - The format interpreter formats/format_sln16.c for the file extension
-   '.sln16' has been removed. The '.sln16' file interpreter now exists
-   in the formats/format_sln.c module along with new support for sln12,
-   sln24, sln32, sln44, sln48, sln96, and sln192 file extensions.
-
-HTTP:
- - A bindaddr must be specified in order for the HTTP server
-   to run. Previous versions would default to 0.0.0.0 if no
-   bindaddr was specified.
-
-Gtalk:
- - The default value for 'context' and 'parkinglots' in gtalk.conf has
-   been changed to 'default', previously they were empty.
-
-chan_dahdi:
- - The mohinterpret=passthrough setting is deprecated in favor of
-   moh_signaling=notify.
-
-pbx_lua:
- - Execution no longer continues after applications that do dialplan jumps
-   (such as app.goto).  Now when an application such as app.goto() is called,
-   control is returned back to the pbx engine and the current extension
-   function stops executing.
- - the autoservice now defaults to being on by default
- - autoservice_start() and autoservice_start() no longer return a value.
-
-Queue:
- - Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
- - QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.
-
-Asterisk Database:
- - The internal Asterisk database has been switched from Berkeley DB 1.86 to
-   SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
-   utility in the UTILS section of menuselect. If an existing astdb is found and no
-   astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
-   convert an existing astdb to the SQLite3 version automatically at runtime. If
-   moving back from Asterisk 10 to Asterisk 1.8, the astdb2bdb utility can be used
-   to create a Berkeley DB copy of the SQLite3 astdb that Asterisk 10 uses.
-
-Manager:
- - The AMI protocol version was incremented to 1.2 as a result of changing two
-   instances of the Unlink event to Bridge events. This change was documented
-   as part of the AMI 1.1 update, but two Unlink events were inadvertently left
-   unchanged.
-
-Module Support Level
- - All modules in the addons, apps, bridge, cdr, cel, channels, codecs,
-   formats, funcs, pbx, and res have been updated to include MODULEINFO data
-   that includes <support_level> tags with a value of core, extended, or deprecated.
-   More information is available on the Asterisk wiki at
-   https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
-
-   Deprecated modules are now marked to not build by default and must be explicitly
-   enabled in menuselect.
-
-chan_sip:
- - Setting of HASH(SIP_CAUSE,<slave-channel-name>) on channels is now disabled
-   by default. It can be enabled using the 'storesipcause' option. This feature
-   has a significant performance penalty.
-
-UDPTL:
- - The default UDPTL port range in udptl.conf.sample differed from the defaults
-   in the source. If you didn't have a config file, you got 4500 to 4599. Now the
-   default is 4000 to 4999.
-
-===========================================================
-===========================================================

UPGRADE-12.txt

@@ -1,478 +0,0 @@
-===========================================================
-===
-=== Information for upgrading between Asterisk versions
-===
-=== These files document all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also include advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
-=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
-=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
-=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
-=== UPGRADE-10.txt  -- Upgrade info for 1.8 to 10
-=== UPGRADE-11.txt  -- Upgrade info for 10 to 11
-===
-===========================================================
-
-There are many significant architectural changes in Asterisk 12. It is
-recommended that you not only read through this document for important
-changes that affect an upgrade, but that you also read through the CHANGES
-document in depth to better understand the new options available to you.
-
-Additional information on the architectural changes made in Asterisk can be
-found on the Asterisk wiki (https://wiki.asterisk.org)
-
-Of particular note, the following systems in Asterisk underwent significant
-changes. Documentation for the changes and a specification for their
-behavior in Asterisk 12 is also available on the Asterisk wiki.
- - AMI: Many events were changed, and the semantics of channels and bridges
-        were defined. In particular, how channels and bridges behave under
-        transfer scenarios and situations involving multiple parties has
-        changed significantly. See https://wiki.asterisk.org/wiki/x/dAFRAQ
-        for more information.
- - CDR: CDR logic was extracted from the many locations it existed in across
-        Asterisk and implemented as a consumer of Stasis message bus events.
-        As a result, consistency of records has improved significantly and the
-        behavior of CDRs in transfer scenarios has been defined in the CDR
-        specification. However, significant behavioral changes in CDRs resulted
-        from the transition. The most significant change is the addition of
-        CDR entries when a channel who is the Party A in a CDR leaves a bridge.
-        See https://wiki.asterisk.org/wiki/x/pwpRAQ for more information.
- - CEL: Much like CDRs, CEL was removed from the many locations it existed in
-        across Asterisk and implemented as a consumer of Stasis message bus
-        events. It now closely follows the Bridging API model of channels and
-        bridges, and has a much closer consistency of conveyed events as AMI.
-        For the changes in events, see https://wiki.asterisk.org/wiki/x/4ICLAQ.
-
-Build System:
- - Removed the CHANNEL_TRACE development mode build option. Certain aspects of
-   the CHANNEL_TRACE build option were incompatible with the new bridging
-   architecture.
-
- - Asterisk now depends on libjansson, libuuid and optionally (but recommended)
-   libxslt and uriparser.
-
- - The new SIP stack and channel driver uses a particular version of PJSIP.
-   Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
-   configuring and installing PJSIP for use with Asterisk.
-
-AgentLogin and chan_agent:
- - Along with AgentRequest, this application has been modified to be a
-   replacement for chan_agent. The chan_agent module and the Agent channel
-   driver have been removed from Asterisk, as the concept of a channel driver
-   proxying in front of another channel driver was incompatible with the new
-   architecture (and has had numerous problems through past versions of
-   Asterisk). The act of a channel calling the AgentLogin application places the
-   channel into a pool of agents that can be requested by the AgentRequest
-   application. Note that this application, as well as all other agent related
-   functionality, is now provided by the app_agent_pool module.
-
- - This application no longer performs agent authentication. If authentication
-   is desired, the dialplan needs to perform this function using the
-   Authenticate or VMAuthenticate application or through an AGI script before
-   running AgentLogin.
-
- - The agents.conf schema has changed. Rather than specifying agents on a
-   single line in comma delineated fashion, each agent is defined in a separate
-   context. This allows agents to use the power of context templates in their
-   definition.
-
- - A number of parameters from agents.conf have been removed. This includes
-   maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
-   urlprefix, and savecallsin. These options were obsoleted by the move from
-   a channel driver model to the bridging/application model provided by
-   app_agent_pool.
-
- - The AGENTUPDATECDR channel variable has also been removed, for the same
-   reason as the updatecdr option.
-
- - The endcall and enddtmf configuration options are removed.  Use the
-   dialplan function CHANNEL(dtmf_features) to set DTMF features on the agent
-   channel before calling AgentLogin.
-
-AgentMonitorOutgoing
- - This application has been removed. It was a holdover from when
-   AgentCallbackLogin was removed.
-
-Answer
- - It is no longer possible to bypass updating the CDR when answering a
-   channel. CDRs are based on the channel state and will be updated when
-   the channel is Answered.
-
-ControlPlayback
- - The channel variable CPLAYBACKSTATUS may now return the value
-   'REMOTESTOPPED' when playback is stopped by an external entity.
-
-DISA
- - This application now has a dependency on the app_cdr module. It uses this
-   module to hide the CDR created prior to execution of the DISA application.
-
-DumpChan:
- - The output of DumpChan no longer includes the DirectBridge or IndirectBridge
-   fields. Instead, if a channel is in a bridge, it includes a BridgeID field
-   containing the unique ID of the bridge that the channel happens to be in.
-
-ForkCDR:
- - Nearly every parameter in ForkCDR has been updated and changed to reflect
-   the changes in CDRs. Please see the documentation for the ForkCDR
-   application, as well as the CDR specification on the Asterisk wiki.
-
-NoCDR:
- - The NoCDR application has been deprecated. Please use the CDR_PROP function
-   to disable CDRs on a channel.
-
-ParkAndAnnounce:
- - The app_parkandannounce module has been removed. The application
-   ParkAndAnnounce is now provided by the res_parking module. See the
-   Parking changes for more information.
-
-ResetCDR:
- - The 'w' and 'a' options have been removed. Dispatching CDRs to registered
-   backends occurs on an as-needed basis in order to preserve linkedid
-   propagation and other needed behavior.
- - The 'e' option is deprecated. Please use the CDR_PROP function to enable
-   CDRs on a channel that they were previously disabled on.
- - The ResetCDR application is no longer a part of core Asterisk, and instead
-   is now delivered as part of app_cdr.
-
-Queues:
- - Queue strategy rrmemory now has a predictable order similar to strategy
-   rrordered. Members will be called in the order that they are added to the
-   queue.
-
- - Removed the queues.conf check_state_unknown option.  It is no longer
-   necessary.
-
- - It is now possible to play the Queue prompts to the first user waiting in a
-   call queue. Note that this may impact the ability for agents to talk with
-   users, as a prompt may still be playing when an agent connects to the user.
-   This ability is disabled by default but can be enabled on an individual
-   queue using the 'announce-to-first-user' option.
-
- - The configuration options eventwhencalled and eventmemberstatus have been
-   removed.  As a result, the AMI events QueueMemberStatus, AgentCalled,
-   AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
-   sent.  The "Variable" fields will also no longer exist on the Agent* events.
-   These events can be filtered out from a connected AMI client using the
-   eventfilter setting in manager.conf.
-
- - The queue log now differentiates between blind and attended transfers. A
-   blind transfer will result in a BLINDTRANSFER message with the destination
-   context and extension. An attended transfer will result in an
-   ATTENDEDTRANSFER message. This message will indicate the method by which
-   the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
-   for running an application on a bridge or channel, or "LINK" for linking
-   two bridges together with local channels. The queue log will also now detect
-   externally initiated blind and attended transfers and record the transfer
-   status accordingly.
-
- - When performing queue pause/unpause on an interface without specifying an
-   individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
-   least one member of any queue exists for that interface.
-
-SetAMAFlags
- - This application is deprecated in favor of CHANNEL(amaflags).
-
-VoiceMail:
- - Mailboxes defined by app_voicemail MUST be referenced by the rest of the
-   system as mailbox@context.  The rest of the system cannot add @default
-   to mailbox identifiers for app_voicemail that do not specify a context
-   any longer.  It is a mailbox identifier format that should only be
-   interpreted by app_voicemail.
-
- - The voicemail.conf configuration file now has an 'alias' configuration
-   parameter for use with the Directory application. The voicemail realtime
-   database table schema has also been updated with an 'alias' column. Systems
-   using voicemail with realtime should update their schemas accordingly.
-
-Channel Drivers:
- - When a channel driver is configured to enable jiterbuffers, they are now
-   applied unconditionally when a channel joins a bridge. If a jitterbuffer
-   is already set for that channel when it enters, such as by the JITTERBUFFER
-   function, then the existing jitterbuffer will be used and the one set by
-   the channel driver will not be applied.
-
-chan_bridge
- - chan_bridge is removed and its functionality is incorporated into ConfBridge
-   itself.
-
-chan_dahdi:
- - Analog port dialing and deferred DTMF dialing for PRI now distinguishes
-   between 'w' and 'W'.  The 'w' pauses dialing for half a second.  The 'W'
-   pauses dialing for one second.
-
- - The default for inband_on_proceeding has changed to no.
-
- - The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
-   A range of channels can be specified to be destroyed. Note that this command
-   should only be used if you understand the risks it entails.
-
- - The script specified by the chan_dahdi.conf mwimonitornotify option now gets
-   the exact configured mailbox name.  For app_voicemail mailboxes this is
-   mailbox@context.
-
- - Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
-
- - ignore_failed_channels now defaults to True: the channel will continue to
-   be configured even if configuring it has failed. This is generally a
-   better setup for systems with not more than one DAHDI device or with DAHDI
-   >= 2.8.0 .
-
-chan_local:
- - The /b option has been removed.
-
- - chan_local moved into the system core and is no longer a loadable module.
-
-chan_sip:
- - The 'callevents' parameter has been removed. Hold AMI events are now raised
-   in the core, and can be filtered out using the 'eventfilter' parameter
-   in manager.conf.
-
- - Dynamic realtime tables for SIP Users can now include a 'path' field. This
-   will store the path information for that peer when it registers. Realtime
-   tables can also use the 'supportpath' field to enable Path header support.
-
- - LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
-   objectIdentifier. This maps to the supportpath option in sip.conf.
-
-Core:
- - Masquerades as an operation inside Asterisk have been effectively hidden
-   by the migration to the Bridging API. As such, many 'quirks' of Asterisk
-   no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
-   dropping of frame/audio hooks, and other internal implementation details
-   that users had to deal with. This fundamental change has large implications
-   throughout the changes documented for this version. For more information
-   about the new core architecture of Asterisk, please see the Asterisk wiki.
-
- - The following channel variables have changed behavior which is described in
-   the CHANGES file: TRANSFER_CONTEXT, BRIDGEPEER, BRIDGEPVTCALLID,
-   ATTENDED_TRANSFER_COMPLETE_SOUND, DYNAMIC_FEATURENAME, and DYNAMIC_PEERNAME.
-
-AMI (Asterisk Manager Interface):
- - Version 1.4 - The details of what happens to a channel when a masquerade
-   happens (transfers, parking, etc) have changed.
-   - The Masquerade event now includes the Uniqueid's of the clone and original
-     channels.
-   - Channels no longer swap Uniqueid's as a result of the masquerade.
-   - Instead of a shell game of renames, there's now a single rename, appending
-     <ZOMBIE> to the name of the original channel.
-
- - *Major* changes were made to both the syntax as well as the semantics of the
-   AMI protocol. In particular, AMI events have been substantially modified
-   and improved in this version of Asterisk. The major event changes are listed
-   below.
-   - NewPeerAccount has been removed. NewAccountCode is raised instead.
-   - Reload events have been consolidated and standardized.
-   - ModuleLoadReport has been removed.
-   - FaxSent is now SendFAX; FaxReceived is now ReceiveFAX. This standardizes
-     app_fax and res_fax events.
-   - MusicOnHold has been replaced with MusicOnHoldStart and MusicOnHoldStop.
-   - JabberEvent has been removed.
-   - Hold is now in the core and will now raise Hold and Unhold events.
-   - Join is now QueueCallerJoin.
-   - Leave is now QueueCallerLeave.
-   - Agentlogin/Agentlogoff is now AgentLogin/AgentLogoff, respectively.
-   - ChannelUpdate has been removed.
-   - Local channel optimization is now conveyed via LocalOptimizationBegin and
-     LocalOptimizationEnd.
-   - BridgeAction and BridgeExec have been removed.
-   - BlindTransfer and AttendedTransfer events were added.
-   - Dial is now DialBegin and DialEnd.
-   - DTMF is now DTMFBegin and DTMFEnd.
-   - Bridge has been replaced with BridgeCreate, BridgeEnter, BridgeLeave, and
-     BridgeDestroy
-   - MusicOnHold has been replaced with MusicOnHoldStart and MusicOnHoldStop
-   - AGIExec is now AGIExecStart and AGIExecEnd
-   - AsyncAGI is now AsyncAGIStart, AsyncAGIExec, and AsyncAGIEnd
-
- - The 'MCID' AMI event now publishes a channel snapshot when available and
-   its non-channel-snapshot parameters now use either the "MCallerID" or
-   'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
-   of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
-   parameters in the channel snapshot.
-
- - The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
-   renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
-
- - All AMI events now contain a 'SystemName' field, if available.
-
- - Local channel information in events is now prefixed with 'LocalOne' and
-   'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
-   the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
-   and 'LocalOptimizationEnd' events.
-
- - The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
-   previous versions. They now report all SR/RR packets sent/received, and
-   have been restructured to better reflect the data sent in a SR/RR. In
-   particular, the event structure now supports multiple report blocks.
-
- - The deprecated use of | (pipe) as a separator in the channelvars setting in
-   manager.conf has been removed.
-
- - The SIP SIPqualifypeer action now sends a response indicating it will qualify
-   a peer once a peer has been found to qualify.  Once the qualify has been
-   completed it will now issue a SIPqualifypeerdone event.
-
- - The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
-   in a future release. Please use the common 'Exten' field instead.
-
- - The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
-   'UnParkedCall' have changed significantly in the new res_parking module.
-   - The 'Channel' and 'From' headers are gone. For the channel that was parked
-     or is coming out of parking, a 'Parkee' channel snapshot is issued and it
-     has a number of fields associated with it. The old 'Channel' header relayed
-     the same data as the new 'ParkeeChannel' header.
-   - The 'From' field was ambiguous and changed meaning depending on the event.
-     for most of these, it was the name of the channel that parked the call
-     (the 'Parker'). There is no longer a header that provides this channel name,
-     however the 'ParkerDialString' will contain a dialstring to redial the
-     device that parked the call.
-   - On UnParkedCall events, the 'From' header would instead represent the
-     channel responsible for retrieving the parkee. It receives a channel
-     snapshot labeled 'Retriever'. The 'from' field is is replaced with
-     'RetrieverChannel'.
-   - Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
-
- - The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
-   fashion has changed the field names 'StartExten' and 'StopExten' to
-   'StartSpace' and 'StopSpace' respectively.
-
- - The AMI 'Status' response event to the AMI Status action replaces the
-   'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
-   indicate what bridge the channel is currently in.
-
-CDR (Call Detail Records)
- - Significant changes have been made to the behavior of CDRs. The CDR engine
-   was effectively rewritten and built on the Stasis message bus. For a full
-   definition of CDR behavior in Asterisk 12, please read the specification
-   on the Asterisk wiki (wiki.asterisk.org).
-
- - CDRs will now be created between all participants in a bridge. For each
-   pair of channels in a bridge, a CDR is created to represent the path of
-   communication between those two endpoints. This lets an end user choose who
-   to bill for what during bridge operations with multiple parties.
-
- - The duration, billsec, start, answer, and end times now reflect the times
-   associated with the current CDR for the channel, as opposed to a cumulative
-   measurement of all CDRs for that channel.
-
-CEL:
- - The Uniqueid field for a channel is now a stable identifier, and will not
-   change due to transfers, parking, etc.
-
- - CEL has undergone significant rework in Asterisk 12, and is now built on the
-   Stasis message bus. Please see the specification for CEL on the Asterisk
-   wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
-   information. A summary of the affected events is below:
-   - BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
-     CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
-     events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT.
-   - BLINDTRANSFER/ATTENDEDTRANSFER events now report the peer as NULL and
-     additional information in the extra string field.
-
-Dialplan Functions:
-
- - Certain dialplan functions have been marked as 'dangerous', and may only be
-   executed from the dialplan. Execution from extenal sources (AMI's GetVar and
-   SetVar actions; etc.) may be inhibited by setting live_dangerously in the
-   [options] section of asterisk.conf to no. SHELL(), channel locking, and
-   direct file read/write functions are marked as dangerous. DB_DELETE() and
-   REALTIME_DESTROY() are marked as dangerous for reads, but can now safely
-   accept writes (which ignore the provided value).
- - The default value for live_dangerously was changed from yes (in Asterisk 11
-   and earlier) to no (in Asterisk 12 and greater).
-
-Dialplan:
- - All channel and global variable names are evaluated in a case-sensitive
-   manner. In previous versions of Asterisk, variables created and evaluated in
-   the dialplan were evaluated case-insensitively, but built-in variables and
-   variable evaluation done internally within Asterisk was done
-   case-sensitively.
-
- - Asterisk has always had code to ignore dash '-' characters that are not
-   part of a character set in the dialplan extensions.  The code now
-   consistently ignores these characters when matching dialplan extensions.
-
- - BRIDGE_FEATURES channel variable is now casesensitive for feature letter
-   codes. Uppercase variants apply them to the calling party while lowercase
-   variants apply them to the called party.
-
-Features:
- - The features.conf [applicationmap] <FeatureName>  ActivatedBy option is
-   no longer honored.  The feature is always activated by the channel that has
-   DYNAMIC_FEATURES defined on it when it enters the bridge. Use predial to set
-   different values of DYNAMIC_FEATURES on the channels
-
- - Executing a dynamic feature on the bridge peer in a multi-party bridge will
-   execute it on all peers of the activating channel.
-
- - There is no longer an explicit 'features reload' CLI command. Features can
-   still be reloaded using 'module reload features'.
-
- - It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
-   features.c for atxferdropcall=no to work properly. This option now just
-   works.
-
-Parking:
- - Parking has been extracted from the Asterisk core as a loadable module,
-   res_parking.
-
- - Configuration is found in res_parking.conf. It is no longer supported in
-   features.conf
-
- - The arguments for the Park, ParkedCall, and ParkAndAnnounce applications
-   have been modified significantly. See the application documents for
-   specific details.
-
- - Numerous changes to Parking related applications, AMI and CLI commands and
-   internal inter-workings  have been made. Please read the CHANGES file for
-   the detailed list.
-
-Security Events Framework:
- - Security Event timestamps now use ISO 8601 formatted date/time instead of
-   the "seconds-microseconds" format that it was using previously.
-
-AGENT:
- - The password option has been disabled, as the AgentLogin application no
-   longer provides authentication.
-
-AUDIOHOOK_INHERIT:
- - Due to changes in the Asterisk core, this function is no longer needed to
-   preserve a MixMonitor on a channel during transfer operations and dialplan
-   execution. It is effectively obsolete.
-
-CDR: (function)
- - The 'amaflags' and 'accountcode' attributes for the CDR function are
-   deprecated. Use the CHANNEL function instead to access these attributes.
-
- - The 'l' option has been removed. When reading a CDR attribute, the most
-   recent record is always used. When writing a CDR attribute, all non-finalized
-   CDRs are updated.
-
- - The 'r' option has been removed, for the same reason as the 'l' option.
-
- - The 's' option has been removed, as LOCKED semantics no longer exist in the
-   CDR engine.
-
-VMCOUNT:
- - Mailboxes defined by app_voicemail MUST be referenced by the rest of the
-   system as mailbox@context.  The rest of the system cannot add @default
-   to mailbox identifiers for app_voicemail that do not specify a context
-   any longer.  It is a mailbox identifier format that should only be
-   interpreted by app_voicemail.
-
-res_rtp_asterisk:
- - ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
-   them, an Asterisk-specific version of PJSIP needs to be installed.
-   Tarballs are available from https://github.com/asterisk/pjproject/tags/.
-
-
-===========================================================
-===========================================================

UPGRADE-13.txt

@@ -1,399 +0,0 @@
-===========================================================
-===
-=== Information for upgrading between Asterisk versions
-===
-=== These files document all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also include advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
-=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
-=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
-=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
-=== UPGRADE-10.txt  -- Upgrade info for 1.8 to 10
-=== UPGRADE-11.txt  -- Upgrade info for 10 to 11
-=== UPGRADE-12.txt  -- Upgrade info for 11 to 12
-===========================================================
-
-General Asterisk Changes:
- - The asterisk command line -I option and the asterisk.conf internal_timing
-   option are removed and always enabled if any timing module is loaded.
-
- - The per console verbose level feature as previously implemented caused a
-   large performance penalty.  The fix required some minor incompatibilities
-   if the new rasterisk is used to connect to an earlier version.  If the new
-   rasterisk connects to an older Asterisk version then the root console verbose
-   level is always affected by the "core set verbose" command of the remote
-   console even though it may appear to only affect the current console.  If
-   an older version of rasterisk connects to the new version then the
-   "core set verbose" command will have no effect.
-
- - The asterisk compatibility options in asterisk.conf have been removed.
-   These options enabled certain backwards compatibility features for
-   pbx_realtime, res_agi, and app_set that made their behaviour similar to
-   Asterisk 1.4. Users who used these backwards compatibility settings should
-   update their dialplans to use ',' instead of '|' as a delimiter, and should
-   use the Set dialplan application instead of the MSet dialplan application.
-
-Build System:
- - Sample config files have been moved from configs/ to a subfolder of that
-   directory, 'samples'.
-
- - The menuselect utility has been pulled into the Asterisk repository. As a
-   result, the libxml2 development library is now a required dependency for
-   Asterisk.
-
- - Added a new Compiler Flag, REF_DEBUG. When enabled, reference counted
-   objects will emit additional debug information to the refs log file located
-   in the standard Asterisk log file directory. This log file is useful in
-   tracking down object leaks and other reference counting issues. Prior to
-   this version, this option was only available by modifying the source code
-   directly. This change also includes a new script, refcounter.py, in the
-   contrib folder that will process the refs log file.
-
-Applications:
-
-ConfBridge:
- - The sound_place_into_conference sound used in Confbridge is now deprecated
-   and is no longer functional since it has been broken since its inception
-   and the fix involved using a different method to achieve the same goal. The
-   new method to achieve this functionality is by using sound_begin to play
-   a sound to the conference when waitmarked users are moved into the conference.
-
- - Added 'Admin' header to ConfbridgeJoin, ConfbridgeLeave, ConfbridgeMute,
-   ConfbridgeUnmute, and ConfbridgeTalking AMI events.
-
-ControlPlayback:
- - The ControlPlayback and 'control stream file' AGI command will no longer
-   implicitly answer the channel. If you do not answer the channel prior to
-   using either this application or AGI command, you must send Progress
-   first.
-
-Queue:
- - Queue rules provided in queuerules.conf can no longer be named "general".
-
-SetMusicOnHold:
- - The SetMusicOnHold dialplan application was deprecated and has been removed.
-   Users of the application should use the CHANNEL function's musicclass
-   setting instead.
-
-WaitMusicOnHold:
- - The WaitMusicOnHold dialplan application was deprecated and has been
-   removed. Users of the application should use MusicOnHold with a duration
-   parameter instead.
-
-CDR Backends:
- - The cdr_sqlite module was deprecated and has been removed. Users of this
-   module should use the cdr_sqlite3_custom module instead.
-
-Channel Drivers:
-
-chan_dahdi:
- - SS7 support now requires libss7 v2.0 or later.
-
- - Added the inband_on_setup_ack compatibility option to chan_dahdi.conf to
-   deal with switches that don't send an inband progress indication in the
-   SETUP ACKNOWLEDGE message.
-   Default is now no.
-
-chan_gtalk
- - This module was deprecated and has been removed. Users of chan_gtalk
-   should use chan_motif.
-
-chan_h323
- - This module was deprecated and has been removed. Users of chan_h323
-   should use chan_ooh323.
-
-chan_jingle
- - This module was deprecated and has been removed. Users of chan_jingle
-   should use chan_motif.
-
-chan_pjsip:
- - Added a 'force_avp' option to chan_pjsip which will force the usage of
-   'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' as the media transport type
-   in SDP offers depending on settings, even when DTLS is used for media
-   encryption.
-
- - Added a 'media_use_received_transport' option to chan_pjsip which will
-   cause the SDP answer to use the media transport as received in the SDP
-   offer.
-
-chan_sip:
- - Made set SIPREFERREDBYHDR as inheritable for better chan_pjsip
-   interoperability.
-
- - The SIPPEER dialplan function no longer supports using a colon as a
-   delimiter for parameters. The parameters for the function should be
-   delimited using a comma.
-
- - The SIPCHANINFO dialplan function was deprecated and has been removed. Users
-   of the function should use the CHANNEL function instead.
-
- - Added a 'force_avp' option for chan_sip. When enabled this option will
-   cause the media transport in the offer or answer SDP to be 'RTP/AVP',
-   'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' even if a DTLS stream has been
-   configured. This option can be set to improve interoperability with WebRTC
-   clients that don't use the RFC defined transport for DTLS.
-
- - The 'dtlsverify' option in chan_sip now has additional values besides
-   'yes' and 'no'. If 'yes' is specified both the certificate and fingerprint
-   will be verified. If 'no' is specified then neither the certificate or
-   fingerprint is verified. If 'certificate' is specified then only the
-   certificate is verified. If 'fingerprint' is specified then only the
-   fingerprint is verified.
-
- - A 'dtlsfingerprint' option has been added to chan_sip which allows the
-   hash to be specified for the DTLS fingerprint placed in SDP. Supported
-   values are 'sha-1' and 'sha-256' with 'sha-256' being the default.
-
- - The 'progressinband=never' option is now more zealous in the persecution of
-   progress messages coming from Asterisk. Channels bridged with a SIP channel
-   that has 'progressinband=never' set will not be able to forward their
-   progress indications through to the SIP device. chan_sip will now turn such
-   progress indications into a 180 Ringing (if a 180 has not yet been
-   transmitted) if 'progressinband=never'.
-
-  - The codec preference order in an SDP during an offer is slightly different
-    than previous releases. Prior to Asterisk 13, the preference order of
-    codecs used to be:
-    (1) Our preferred codec
-    (2) Our configured codecs
-    (3) Any non-audio joint codecs
-
-    One of the ways the new media format architecture in Asterisk 13 improves
-    performance is by reference counting formats, such that they can be reused
-    in many places without additional allocation. To not require a large
-    amount of locking, an instance of a format is immutable by convention.
-    This works well except for formats with attributes. Since a media format
-    with an attribute is a different object than the same format without an
-    attribute, we have to carry over the formats with attributes from an
-    inbound offer so that the correct attributes are offered in an outgoing
-    INVITE request. This requires some subtle tweaks to the preference order
-    to ensure that the media format with attributes is offered to a remote
-    peer, as opposed to the same media format (but without attributes) that
-    may be stored in the peer object.
-
-    All of this means that our offer offer list will now be:
-    (1) Our preferred codec
-    (2) Any joint codecs offered by the inbound offer
-    (3) All other codecs that are not the preferred codec and not a joint
-        codec offered by the inbound offer
-
-chan_unistim:
- - The unistim.conf 'dateformat' has changed meaning of options values to conform
-   values used inside Unistim protocol
-
- - Added 'dtmf_duration' option with changing default operation to disable
-   receivied dtmf playback on unistim phone
-
-Core:
-
-Account Codes:
- - accountcode behavior changed somewhat to add functional peeraccount
-   support.  The main change is that local channels now cross accountcode
-   and peeraccount across the special bridge between the ;1 and ;2 channels
-   just like channels between normal bridges.  See the CHANGES file for
-   more information.
-
-ARI:
- - The ARI version has been changed to 1.5.0. This is to reflect backwards
-   compatible changes made since 12.0.0 was released.
-
- - Added a new ARI resource 'mailboxes' which allows the creation and
-   modification of mailboxes managed by external MWI. Modules res_mwi_external
-   and res_stasis_mailbox must be enabled to use this resource.
-
- - Added new events for externally initiated transfers. The event
-   BridgeBlindTransfer is now raised when a channel initiates a blind transfer
-   of a bridge in the ARI controlled application to the dialplan; the
-   BridgeAttendedTransfer event is raised when a channel initiates an
-   attended transfer of a bridge in the ARI controlled application to the
-   dialplan.
-
- - Channel variables may now be specified as a body parameter to the
-   POST /channels operation. The 'variables' key in the JSON is interpreted
-   as a sequence of key/value pairs that will be added to the created channel
-   as channel variables. Other parameters in the JSON body are treated as
-   query parameters of the same name.
-
- - A bug fix in bridge creation has caused a behavioural change in how
-   subscriptions are created for bridges. A bridge created through ARI, does
-   not, by itself, have a subscription created for any particular Stasis
-   application. When a channel in a Stasis application joins a bridge, an
-   implicit event subscription is created for that bridge as well. Previously,
-   when a channel left such a bridge, the subscription was leaked; this allowed
-   for later bridge events to continue to be pushed to the subscribed
-   applications. That leak has been fixed; as a result, bridge events that were
-   delivered after a channel left the bridge are no longer delivered. An
-   application must subscribe to a bridge through the applications resource if
-   it wishes to receive all events related to a bridge.
-
-AMI:
- - The AMI version has been changed to 2.5.0. This is to reflect backwards
-   compatible changes made since 12.0.0 was released.
-
- - The DialStatus field in the DialEnd event can now have additional values.
-   This includes ABORT, CONTINUE, and GOTO.
-
- - The res_mwi_external_ami module can, if loaded, provide additional AMI
-   actions and events that convey MWI state within Asterisk. This includes
-   the MWIGet, MWIUpdate, and MWIDelete actions, as well as the MWIGet and
-   MWIGetComplete events that occur in response to an MWIGet action.
-
- - AMI now contains a new class authorization, 'security'. This is used with
-   the following new events: FailedACL, InvalidAccountID, SessionLimit,
-   MemoryLimit, LoadAverageLimit, RequestNotAllowed, AuthMethodNotAllowed,
-   RequestBadFormat, SuccessfulAuth, UnexpectedAddress, ChallengeResponseFailed,
-   InvalidPassword, ChallengeSent, and InvalidTransport.
-
- - Bridge related events now have two additional fields: BridgeName and
-   BridgeCreator. BridgeName is a descriptive name for the bridge;
-   BridgeCreator is the name of the entity that created the bridge. This
-   affects the following events: ConfbridgeStart, ConfbridgeEnd,
-   ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
-   ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
-   AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
-
- - MixMonitor AMI actions now require users to have authorization classes.
-   * MixMonitor - system
-   * MixMonitorMute - call or system
-   * StopMixMonitor - call or system
-
- - Removed the undocumented manager.conf block-sockets option.  It interferes with
-   TCP/TLS inactivity timeouts.
-
- - The response to the PresenceState AMI action has historically contained two
-   Message keys. The first of these is used as an informative message regarding
-   the success/failure of the action; the second contains a Presence state
-   specific message. Having two keys with the same unique name in an AMI
-   message is cumbersome for some client; hence, the Presence specific Message
-   has been deprecated. The message will now contain a PresenceMessage key
-   for the presence specific information; the Message key containing presence
-   information will be removed in the next major version of AMI.
-
- - The manager.conf 'eventfilter' now takes an "extended" regular expression
-   instead of a "basic" one.
-
-CDRs:
- - The "endbeforehexten" setting now defaults to "yes", instead of "no".
-   When set to "no", yhis setting will cause a new CDR to be generated when a
-   channel enters into hangup logic (either the 'h' extension or a hangup
-   handler subroutine). In general, this is not the preferred default: this
-   causes extra CDRs to be generated for a channel in many common dialplans.
-
-CLI commands:
- - "core show settings" now lists the current console verbosity in addition
-   to the root console verbosity.
-
- - "core set verbose" has not been able to support the by module verbose
-   logging levels since verbose logging levels were made per console.  That
-   syntax is now removed and a silence option added in its place.
-
-Logging:
- - The 'verbose' setting in logger.conf still takes an optional argument,
-   specifying the verbosity level for each logging destination.  However,
-   the default is now to once again follow the current root console level.
-   As a result, using the AMI Command action with "core set verbose" could
-   again set the root console verbose level and affect the verbose level
-   logged.
-
-HTTP:
- - Added http.conf session_inactivity timer option to close HTTP connections
-   that aren't doing anything.
-
- - Added support for persistent HTTP connections.  To enable persistent
-   HTTP connections configure the keep alive time between HTTP requests.  The
-   keep alive time between HTTP requests is configured in http.conf with the
-   session_keep_alive parameter.
-
-Realtime Configuration:
- - WARNING: The database migration script that adds the 'extensions' table for
-   realtime had to be modified due to an error when installing for MySQL.  The
-   'extensions' table's 'id' column was changed to be a primary key.  This could
-   potentially cause a migration problem.  If so, it may be necessary to
-   manually alter the affected table/column to bring it back in line with the
-   migration scripts.
-
- - New columns have been added to realtime tables for 'support_path' on
-   ps_registrations and ps_aors and for 'path' on ps_contacts for the new
-   SIP Path support in chan_pjsip.
-
- - The following new tables have been added for pjsip realtime: 'ps_systems',
-   'ps_globals', 'ps_tranports', 'ps_registrations'.
-
- - The following columns were added to the 'ps_aors' realtime table:
-   'maximum_expiration', 'outbound_proxy', and 'support_path'.
-
- - The following columns were added to the 'ps_contacts' realtime table:
-   'outbound_proxy', 'user_agent', and 'path'.
-
- - New columns have been added to the ps_endpoints realtime table for the
-   'media_address', 'redirect_method' and 'set_var' options.  Also the
-   'mwi_fromuser' column was renamed to 'mwi_from_user'. A new column
-   'message_context' was added to let users configure how MESSAGE requests are
-   routed to the dialplan.
-
- - A new column was added to the 'ps_globals' realtime table for the 'debug'
-   option.
-
- - PJSIP endpoint columns 'tos_audio' and 'tos_video' have been changed from
-   yes/no enumerators to string values. 'cos_audio' and 'cos_video' have been
-   changed from yes/no enumerators to integer values. PJSIP transport column
-   'tos' has been changed from a yes/no enumerator to a string value. 'cos' has
-   been changed from a yes/no enumerator to an integer value.
-
- - The 'queues' and 'queue_members' realtime tables have been added to the
-   config Alembic scripts.
-
- - A new set of Alembic scripts has been added for CDR tables. This will create
-   a 'cdr' table with the default schema that Asterisk expects.
-
- - A new upgrade script has been added that adds a 'queue_rules' table for
-   app_queue. Users of app_queue can store queue rules in a database. It is
-   important to note that app_queue only looks for this table on module load or
-   module reload; for more information, see the CHANGES file.
-
-Resources:
-
-res_odbc:
-- The compatibility setting, allow_empty_string_in_nontext, has been removed.
-  Empty column values will be stored as empty strings during realtime updates.
-
-res_jabber:
- - This module was deprecated and has been removed. Users of this module should
-   use res_xmpp instead.
-
-res_http_websocket:
- - Added a compatibility option to ari.conf, sip.conf, and pjsip.conf
-   'websocket_write_timeout'. When a websocket connection exists where Asterisk
-   writes a substantial amount of data to the connected client, and the connected
-   client is slow to process the received data, the socket may be disconnected.
-   In such cases, it may be necessary to adjust this value.
-   Default is 100 ms.
-Scripts:
-
-safe_asterisk:
- - The safe_asterisk script was previously not installed on top of an existing
-   version. This caused bug-fixes in that script not to be deployed. If your
-   safe_asterisk script is customized, be sure to keep your changes. Custom
-   values for variables should be created in *.sh file(s) inside
-   ASTETCDIR/startup.d/. See ASTERISK-21965.
-
- - Changed a log message in safe_asterisk and the $NOTIFY mail subject. If
-   you use tools to parse either of them, update your parse functions
-   accordingly. The changed strings are:
-   - "Exited on signal $EXITSIGNAL" => "Asterisk exited on signal $EXITSIGNAL."
-   - "Asterisk Died" => "Asterisk on $MACHINE died (sig $EXITSIGNAL)"
-
-Utilities:
- - The refcounter program has been removed in favor of the refcounter.py script
-   in contrib/scripts.
-
-===========================================================
-===========================================================

UPGRADE-14.txt

@@ -1,115 +0,0 @@
-===========================================================
-===
-=== Information for upgrading between Asterisk versions
-===
-=== These files document all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also include advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
-=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
-=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
-=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
-=== UPGRADE-10.txt  -- Upgrade info for 1.8 to 10
-=== UPGRADE-11.txt  -- Upgrade info for 10 to 11
-=== UPGRADE-12.txt  -- Upgrade info for 11 to 12
-=== UPGRADE-13.txt  -- Upgrade info for 12 to 13
-===========================================================
-
-From 14.6.0 to 14.7.0:
-
-Core:
- - ast_app_parse_timelen now returns an error if it encounters extra characters
-   at the end of the string to be parsed.
-
-From 14.4.0 to 14.5.0:
-
-Core:
- - Support for embedded modules has been removed.  This has not worked in
-   many years.  LOADABLE_MODULES menuselect option is also removed as
-   loadable module support is now always enabled.
-
-From 14.3.0 to 14.4.0:
-
-res_rtp_asterisk:
- - The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
-   Data and Control Packets on a Single Port." For the PJSIP channel driver,
-   chan_pjsip, you can set "rtcp_mux = yes" on a PJSIP endpoint in pjsip.conf
-   to enable the feature. For chan_sip you can set "rtcp_mux = yes" either
-   globally or on a per-peer basis in sip.conf.
-
-New in 14.0.0
-
-ARI:
- - The policy for when to send "Dial" events has changed. Previously, "Dial"
-   events were sent on the calling channel's topic. However, starting in Asterisk
-   14, if there is no calling channel on which to send the event, the event is
-   instead sent on the called channel's topic. Note that for the ARI channels
-   resource's dial operation, this means that the "Dial" events will always be
-   sent on the called channel's topic.
-
-Channel Drivers:
-
-chan_dahdi:
- - For users using the FXO port (FXS signaling) distinctive ring detection
-   feature, you will need to adjust the dringX count values.  The count
-   values now only record ring end events instead of any DAHDI event.  A
-   ring-ring-ring pattern would exceed the pattern limits and stop
-   Caller-ID detection.
-
-chan_sip:
- - The SIP dial string has been extended past the [!dnid] option by another
-   exclamation mark: [!dnid[!fromuri].  An exclamation mark in the To-URI
-   will now mean changes to the From-URI.
-
-Core:
- - The REF_DEBUG compiler flag is now used to enable refdebug by default.
-   The setting can be overridden in asterisk.conf by setting refdebug in
-   the options category.  No recompile is required to enable/disable it.
-
- - Modified processing of command-line options to first parse only what
-   is necessary to read asterisk.conf. Once asterisk.conf is fully loaded,
-   the remaining options are processed.  The -X option now applies to
-   asterisk.conf only.  To enable #exec for other config files you must
-   set execincludes=yes in asterisk.conf.  Any other option set on the
-   command-line will now override the equivalent setting from asterisk.conf.
-
-AMI:
- - The 'ModuleCheck' Action's Version key will no longer show the module
-   version. The value will always be blank.
-
-CLI:
- - The 'core show file version' command has been removed. When Asterisk
-   moved to Git, the source control version support was removed. As a
-   result, the CLi command was no longer useful and was removed as well.
-
-Logging:
- - The first callid created is now 1 instead of 0.  The value 0
-   is now reserved to represent a lack of callid.
-
-AMI:
- - The Command action now sends the output from the CLI command as a series
-   of Output headers for each line instead of as a block of text with the
-   --END COMMAND-- delimiter to match the output from other actions.
-
-   Commands that fail to execute (no such command, invalid syntax etc.) now
-   return an Error response instead of Success.
-
-app_amd:
- - The 'maximum_number_of_words' configuration option and parameter to the AMD
-   application previously did not match the documented functionality + variable
-   name.  In Asterisk 13, a value of '3' would mean that if '3' words were detected,
-   the result would be detection as a 'MACHINE'.  As of this version, the value
-   reflects the maximum words that if EXCEEDED (rather than reached), would
-   result in detection as a machine.  This means that you should update this
-   value to be one higher than your previos value, if your previous value
-   was working well for you.
-
-===========================================================
-===========================================================

UPGRADE-15.txt

@@ -1,63 +0,0 @@
-===========================================================
-===
-=== Information for upgrading between Asterisk versions
-===
-=== These files document all the changes that MUST be taken
-=== into account when upgrading between the Asterisk
-=== versions listed below. These changes may require that
-=== you modify your configuration files, dialplan or (in
-=== some cases) source code if you have your own Asterisk
-=== modules or patches. These files also include advance
-=== notice of any functionality that has been marked as
-=== 'deprecated' and may be removed in a future release,
-=== along with the suggested replacement functionality.
-===
-=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
-=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
-=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
-=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
-=== UPGRADE-10.txt  -- Upgrade info for 1.8 to 10
-=== UPGRADE-11.txt  -- Upgrade info for 10 to 11
-=== UPGRADE-12.txt  -- Upgrade info for 11 to 12
-=== UPGRADE-13.txt  -- Upgrade info for 12 to 13
-=== UPGRADE-14.txt  -- Upgrade info for 13 to 14
-===========================================================
-
-From 15.2.0 to 15.3.0:
-
-res_pjsip
-------------------
- * Users who are matching endpoints by SIP header need to reevaluate their
-   global "endpoint_identifier_order" option in light of the "ip" endpoint
-   identifier method split into the "ip" and "header" endpoint identifier
-   methods.
-
-res_pjsip_endpoint_identifier_ip
-------------------
- * The endpoint identifier "ip" method previously recognized endpoints either
-   by IP address or a matching SIP header.  The "ip" endpoint identifier method
-   is now split into the "ip" and "header" endpoint identifier methods.  The
-   "ip" endpoint identifier method only matches by IP address and the "header"
-   endpoint identifier method only matches by SIP header.  The split allows the
-   user to control the relative priority of the IP address and the SIP header
-   identification methods in the global "endpoint_identifier_order" option.
-   e.g., If you have two type=identify sections where one matches by IP address
-   for endpoint alice and the other matches by SIP header for endpoint bob then
-   you can now predict which endpoint is matched when a request comes in that
-   matches both.
-
-New in 15.0.0:
-
-Build System:
- - '--with-pjproject-bundled' is now the default when running ./configure
-   It can be disabled with '--without-pjproject-bundled'.
-
-Core:
- - Multi-stream support has been added so a channel can have multiple
-   streams of the same type such as audio and video.
-
- - The 'Data Retrieval API' has been removed. This API was not actively
-   maintained, was not added to new modules (such as res_pjsip), and there
-   exist better alternatives to acquire the same information, such as the
-   ARI. As a result, the 'DataGet' AMI action as well as the 'data get'
-   CLI command have been removed.

addons/Makefile

@@ -64,6 +64,10 @@ chan_ooh323.so: _ASTCFLAGS+=$(H323CFLAGS)
 $(call MOD_ADD_C,chan_ooh323,$(H323SOURCE))
 
 ifneq ($(wildcard mp3/Makefile),)
+# At the current time, the fate of mp3 is in flux so it didn't make sense to
+# add configure/makeopts processing for array-bounds since this is the only
+# source file that needs that warning suppressed.
+mp3/layer3.o: _ASTCFLAGS+=-Wno-array-bounds
 $(call MOD_ADD_C,format_mp3,mp3/common.c mp3/dct64_i386.c mp3/decode_ntom.c mp3/layer3.c mp3/tabinit.c mp3/interface.c)
 
 .PHONY: check_mp3

addons/app_mysql.c

@@ -37,6 +37,8 @@
 	<defaultenabled>no</defaultenabled>
 	<support_level>deprecated</support_level>
 	<replacement>func_odbc</replacement>
+	<deprecated_in>1.8</deprecated_in>
+	<removed_in>19</removed_in>
  ***/
 
 #include "asterisk.h"

addons/cdr_mysql.c

@@ -38,6 +38,8 @@
 	<defaultenabled>no</defaultenabled>
 	<support_level>deprecated</support_level>
 	<replacement>cdr_adaptive_odbc</replacement>
+	<deprecated_in>1.8</deprecated_in>
+	<removed_in>19</removed_in>
  ***/
 
 #include "asterisk.h"
@@ -173,7 +175,9 @@ static int mysql_log(struct ast_cdr *cdr)
 {
 	struct ast_str *sql1 = ast_str_thread_get(&sql1_buf, 1024), *sql2 = ast_str_thread_get(&sql2_buf, 1024);
 	int retries = 5;
-#if MYSQL_VERSION_ID >= 50013
+#ifdef HAVE_MYSQLCLIENT_BOOL
+	bool my_bool_true = 1;
+#elif HAVE_MYSQLCLIENT_MY_BOOL
 	my_bool my_bool_true = 1;
 #endif
 
@@ -382,6 +386,13 @@ static int my_unload_module(int reload)
 {
 	struct column *entry;
 
+	if (!reload) {
+		if (ast_cdr_unregister(name)) {
+			/* If we can't unregister the backend, we can't unload the module */
+			return -1;
+		}
+	}
+
 	ast_cli_unregister_multiple(cdr_mysql_status_cli, sizeof(cdr_mysql_status_cli) / sizeof(struct ast_cli_entry));
 
 	if (connected) {
@@ -406,7 +417,8 @@ static int my_unload_module(int reload)
 	if (reload) {
 		return ast_cdr_backend_suspend(name);
 	} else {
-		return ast_cdr_unregister(name);
+		/* We unregistered earlier */
+		return 0;
 	}
 }
 
@@ -460,7 +472,9 @@ static int my_connect_db(struct ast_config *cfg)
 	MYSQL_ROW row;
 	MYSQL_RES *result;
 	char sqldesc[128];
-#if MYSQL_VERSION_ID >= 50013
+#ifdef HAVE_MYSQLCLIENT_BOOL
+	bool my_bool_true = 1;
+#elif HAVE_MYSQLCLIENT_MY_BOOL
 	my_bool my_bool_true = 1;
 #endif
 

addons/chan_mobile.c

@@ -102,7 +102,7 @@ struct adapter_pvt {
 	char id[31];					/* the 'name' from mobile.conf */
 	bdaddr_t addr;					/* adddress of adapter */
 	unsigned int inuse:1;				/* are we in use ? */
-	unsigned int alignment_detection:1;		/* do alignment detection on this adpater? */
+	unsigned int alignment_detection:1;		/* do alignment detection on this adapter? */
 	struct io_context *io;				/*!< io context for audio connections */
 	struct io_context *accept_io;			/*!< io context for sco listener */
 	int *sco_id;					/*!< the io context id of the sco listener socket */
@@ -132,7 +132,8 @@ struct mbl_pvt {
 	int rfcomm_socket;				/* rfcomm socket descriptor */
 	char rfcomm_buf[256];
 	char io_buf[CHANNEL_FRAME_SIZE + AST_FRIENDLY_OFFSET];
-	struct ast_smoother *smoother;			/* our smoother, for making 48 byte frames */
+	struct ast_smoother *bt_out_smoother;			/* our bt_out_smoother, for making 48 byte frames */
+	struct ast_smoother *bt_in_smoother;			/* our smoother, for making "normal" CHANNEL_FRAME_SIZEed byte frames */
 	int sco_socket;					/* sco socket descriptor */
 	pthread_t monitor_thread;			/* monitor thread handle */
 	int timeout;					/*!< used to set the timeout for rfcomm data (may be used in the future) */
@@ -856,7 +857,8 @@ static struct ast_channel *mbl_new(int state, struct mbl_pvt *pvt, struct cidinf
 	else
 		pvt->do_alignment_detection = 0;
 
-	ast_smoother_reset(pvt->smoother, DEVICE_FRAME_SIZE);
+	ast_smoother_reset(pvt->bt_out_smoother, DEVICE_FRAME_SIZE);
+	ast_smoother_reset(pvt->bt_in_smoother, CHANNEL_FRAME_SIZE);
 	ast_dsp_digitreset(pvt->dsp);
 
 	chn = ast_channel_alloc(1, state,
@@ -1131,23 +1133,27 @@ static struct ast_frame *mbl_read(struct ast_channel *ast)
 	pvt->fr.delivery.tv_usec = 0;
 	pvt->fr.data.ptr = pvt->io_buf + AST_FRIENDLY_OFFSET;
 
-	if ((r = read(pvt->sco_socket, pvt->fr.data.ptr, DEVICE_FRAME_SIZE)) == -1) {
-		if (errno != EAGAIN && errno != EINTR) {
-			ast_debug(1, "[%s] read error %d, going to wait for new connection\n", pvt->id, errno);
-			close(pvt->sco_socket);
-			pvt->sco_socket = -1;
-			ast_channel_set_fd(ast, 0, -1);
+	do {
+		if ((r = read(pvt->sco_socket, pvt->fr.data.ptr, DEVICE_FRAME_SIZE)) == -1) {
+			if (errno != EAGAIN && errno != EINTR) {
+				ast_debug(1, "[%s] read error %d, going to wait for new connection\n", pvt->id, errno);
+				close(pvt->sco_socket);
+				pvt->sco_socket = -1;
+				ast_channel_set_fd(ast, 0, -1);
+			}
+			goto e_return;
 		}
-		goto e_return;
-	}
 
-	pvt->fr.datalen = r;
-	pvt->fr.samples = r / 2;
+		pvt->fr.datalen = r;
+		pvt->fr.samples = r / 2;
 
-	if (pvt->do_alignment_detection)
-		do_alignment_detection(pvt, pvt->fr.data.ptr, r);
+		if (pvt->do_alignment_detection)
+			do_alignment_detection(pvt, pvt->fr.data.ptr, r);
 
-	fr = ast_dsp_process(ast, pvt->dsp, &pvt->fr);
+		ast_smoother_feed(pvt->bt_in_smoother, &pvt->fr);
+		fr = ast_smoother_read(pvt->bt_in_smoother);
+	} while (fr == NULL);
+	fr = ast_dsp_process(ast, pvt->dsp, fr);
 
 	ast_mutex_unlock(&pvt->lock);
 
@@ -1174,9 +1180,9 @@ static int mbl_write(struct ast_channel *ast, struct ast_frame *frame)
 		CHANNEL_DEADLOCK_AVOIDANCE(ast);
 	}
 
-	ast_smoother_feed(pvt->smoother, frame);
+	ast_smoother_feed(pvt->bt_out_smoother, frame);
 
-	while ((f = ast_smoother_read(pvt->smoother))) {
+	while ((f = ast_smoother_read(pvt->bt_out_smoother))) {
 		sco_write(pvt->sco_socket, f->data.ptr, f->datalen);
 	}
 
@@ -2000,7 +2006,7 @@ static int at_match_prefix(char *buf, char *prefix)
 }
 
 /*!
- * \brief Read an AT message and clasify it.
+ * \brief Read an AT message and classify it.
  * \param rsock an rfcomm socket
  * \param buf the buffer to store the result in
  * \param count the size of the buffer or the maximum number of characters to read
@@ -2150,10 +2156,12 @@ static inline const char *at_msg2str(at_message_t msg)
  * \param buf the buffer to parse (null terminated)
  * \return -1 on error (parse error) or a ECAM value on success
  *
- * Example string: *ECAV: <ccid>,<ccstatus>,<calltype>[,<processid>]
- * [,exitcause][,<number>,<type>]
+ * Example:
+ * \verbatim *ECAV: <ccid>,<ccstatus>,<calltype>[,<processid>]
+                    [,exitcause][,<number>,<type>] \endverbatim
  *
- * Example indicating busy: *ECAV: 1,7,1
+ * Example indicating busy:
+ * \verbatim *ECAV: 1,7,1 \endverbatim
  */
 static int hfp_parse_ecav(struct hfp_pvt *hfp, char *buf)
 {
@@ -2170,7 +2178,7 @@ static int hfp_parse_ecav(struct hfp_pvt *hfp, char *buf)
 }
 
 /*!
- * \brief Enable Sony Erricson extensions / indications.
+ * \brief Enable Sony Ericsson extensions / indications.
  * \param hfp an hfp_pvt struct
  */
 static int hfp_send_ecam(struct hfp_pvt *hfp)
@@ -3034,7 +3042,7 @@ static void msg_queue_free_and_pop(struct mbl_pvt *pvt)
 }
 
 /*!
- * \brief Remove all itmes from the queue and free them.
+ * \brief Remove all items from the queue and free them.
  * \param pvt a mbl_pvt structure
  */
 static void msg_queue_flush(struct mbl_pvt *pvt)
@@ -3341,7 +3349,7 @@ static int handle_response_ok(struct mbl_pvt *pvt, char *buf)
 			}
 			break;
 		case AT_CLIP:
-			ast_debug(1, "[%s] caling line indication enabled\n", pvt->id);
+			ast_debug(1, "[%s] calling line indication enabled\n", pvt->id);
 			if (hfp_send_ecam(pvt->hfp) || msg_queue_push(pvt, AT_OK, AT_ECAM)) {
 				ast_debug(1, "[%s] error enabling Sony Ericsson call monitoring extensions\n", pvt->id);
 				goto e_return;
@@ -3561,7 +3569,7 @@ static int handle_response_ciev(struct mbl_pvt *pvt, char *buf)
 			if (pvt->owner) {
 				ast_debug(1, "[%s] hanging up owner\n", pvt->id);
 				if (mbl_queue_hangup(pvt)) {
-					ast_log(LOG_ERROR, "[%s] error queueing hangup, disconnectiong...\n", pvt->id);
+					ast_log(LOG_ERROR, "[%s] error queueing hangup, disconnecting...\n", pvt->id);
 					return -1;
 				}
 			}
@@ -3595,7 +3603,7 @@ static int handle_response_ciev(struct mbl_pvt *pvt, char *buf)
 						handle_response_busy(pvt);
 					}
 					if (mbl_queue_hangup(pvt)) {
-						ast_log(LOG_ERROR, "[%s] error queueing hangup, disconnectiong...\n", pvt->id);
+						ast_log(LOG_ERROR, "[%s] error queueing hangup, disconnecting...\n", pvt->id);
 						return -1;
 					}
 				}
@@ -4425,7 +4433,7 @@ static struct adapter_pvt *mbl_load_adapter(struct ast_config *cfg, const char *
 
 	/* bind the sco listener socket */
 	if (sco_bind(adapter) < 0) {
-		ast_log(LOG_ERROR, "Skipping adapter %s. Error binding audio connection listerner socket.\n", adapter->id);
+		ast_log(LOG_ERROR, "Skipping adapter %s. Error binding audio connection listener socket.\n", adapter->id);
 		goto e_destroy_io;
 	}
 
@@ -4437,7 +4445,7 @@ static struct adapter_pvt *mbl_load_adapter(struct ast_config *cfg, const char *
 
 	/* start the sco listener for this adapter */
 	if (ast_pthread_create_background(&adapter->sco_listener_thread, NULL, do_sco_listen, adapter)) {
-		ast_log(LOG_ERROR, "Skipping adapter %s. Error creating audio connection listerner thread.\n", adapter->id);
+		ast_log(LOG_ERROR, "Skipping adapter %s. Error creating audio connection listener thread.\n", adapter->id);
 		goto e_remove_sco;
 	}
 
@@ -4493,7 +4501,7 @@ static struct mbl_pvt *mbl_load_device(struct ast_config *cfg, const char *cat)
 	}
 	AST_RWLIST_UNLOCK(&adapters);
 	if (!adapter) {
-		ast_log(LOG_ERROR, "Skiping device %s. Unknown adapter '%s' specified.\n", cat, adapter_str);
+		ast_log(LOG_ERROR, "Skipping device %s. Unknown adapter '%s' specified.\n", cat, adapter_str);
 		goto e_return;
 	}
 
@@ -4530,16 +4538,22 @@ static struct mbl_pvt *mbl_load_device(struct ast_config *cfg, const char *cat)
 	pvt->ring_sched_id = -1;
 	pvt->has_sms = 1;
 
-	/* setup the smoother */
-	if (!(pvt->smoother = ast_smoother_new(DEVICE_FRAME_SIZE))) {
-		ast_log(LOG_ERROR, "Skipping device %s. Error setting up frame smoother.\n", cat);
+	/* setup the bt_out_smoother */
+	if (!(pvt->bt_out_smoother = ast_smoother_new(DEVICE_FRAME_SIZE))) {
+		ast_log(LOG_ERROR, "Skipping device %s. Error setting up frame bt_out_smoother.\n", cat);
 		goto e_free_pvt;
 	}
 
+	/* setup the bt_in_smoother */
+	if (!(pvt->bt_in_smoother = ast_smoother_new(CHANNEL_FRAME_SIZE))) {
+		ast_log(LOG_ERROR, "Skipping device %s. Error setting up frame bt_in_smoother.\n", cat);
+		goto e_free_bt_out_smoother;
+	}
+
 	/* setup the dsp */
 	if (!(pvt->dsp = ast_dsp_new())) {
 		ast_log(LOG_ERROR, "Skipping device %s. Error setting up dsp for dtmf detection.\n", cat);
-		goto e_free_smoother;
+		goto e_free_bt_in_smoother;
 	}
 
 	/* setup the scheduler */
@@ -4599,8 +4613,10 @@ e_free_sched:
 	ast_sched_context_destroy(pvt->sched);
 e_free_dsp:
 	ast_dsp_free(pvt->dsp);
-e_free_smoother:
-	ast_smoother_free(pvt->smoother);
+e_free_bt_in_smoother:
+	ast_smoother_free(pvt->bt_in_smoother);
+e_free_bt_out_smoother:
+	ast_smoother_free(pvt->bt_out_smoother);
 e_free_pvt:
 	ast_free(pvt);
 e_return:
@@ -4732,7 +4748,8 @@ static int unload_module(void)
 			ast_free(pvt->hfp);
 		}
 
-		ast_smoother_free(pvt->smoother);
+		ast_smoother_free(pvt->bt_out_smoother);
+		ast_smoother_free(pvt->bt_in_smoother);
 		ast_dsp_free(pvt->dsp);
 		ast_sched_context_destroy(pvt->sched);
 		ast_free(pvt);

addons/chan_ooh323.c

@@ -82,7 +82,7 @@
 /* Defaults */
 #define DEFAULT_CONTEXT "default"
 #define DEFAULT_H323ID "Asterisk PBX"
-#define DEFAULT_LOGFILE "/var/log/asterisk/h323_log"
+#define DEFAULT_LOGFILE "h323_log"
 #define DEFAULT_H323ACCNT "ast_h323"
 
 /* Flags */
@@ -346,7 +346,8 @@ void onModeChanged(ooCallData *call, int t38mode);
 
 extern OOH323EndPoint gH323ep;
 
-static char gLogFile[256] = DEFAULT_LOGFILE;
+static char gLogFile[PATH_MAX] = DEFAULT_LOGFILE;
+static char gInitError[256] = "";
 static int  gPort = 1720;
 static char gIP[2+8*4+7];	/* Max for IPv6 addr */
 struct ast_sockaddr bindaddr;
@@ -2059,7 +2060,7 @@ int onOutgoingCall(ooCallData *call)
 			}
 			ooCallAddAliasDialedDigits(call, p->caller_dialedDigits);
 		} else if (!ast_strlen_zero(p->callerid_num)) {
-			if (ooIsDailedDigit(p->callerid_num)) {
+			if (ooIsDialedDigit(p->callerid_num)) {
 				if (gH323Debug) {
 					ast_verb(0, "setting callid number %s\n", p->callerid_num);
 				}
@@ -2140,7 +2141,7 @@ int onNewCallCreated(ooCallData *call)
 			}
 			ooCallAddAliasDialedDigits(call, p->caller_dialedDigits);
 		} else if (!ast_strlen_zero(p->callerid_num)) {
-			if (ooIsDailedDigit(p->callerid_num)) {
+			if (ooIsDialedDigit(p->callerid_num)) {
 				if (gH323Debug) {
 					ast_verb(0, "setting callid number %s\n", p->callerid_num);
 				}
@@ -2152,7 +2153,7 @@ int onNewCallCreated(ooCallData *call)
 
 
 		if (!ast_strlen_zero(p->exten))  {
-			if (ooIsDailedDigit(p->exten)) {
+			if (ooIsDialedDigit(p->exten)) {
 				ooCallSetCalledPartyNumber(call, p->exten);
 				ooCallAddRemoteAliasDialedDigits(call, p->exten);
 			} else {
@@ -2856,7 +2857,7 @@ int reload_config(int reload)
 	}
 
 	/* Inintialize everything to default */
-	strcpy(gLogFile, DEFAULT_LOGFILE);
+	snprintf(gLogFile, sizeof(gLogFile), "%s/%s", ast_config_AST_LOG_DIR, DEFAULT_LOGFILE);
 	gPort = 1720;
 	gIP[0] = '\0';
 	strcpy(gCallerID, DEFAULT_H323ID);
@@ -3034,7 +3035,11 @@ int reload_config(int reload)
 		        ast_copy_string(gRASIP, v->value, sizeof(gRASIP));
 			ast_verb(3, "  == Setting RAS IP to %s\n", gRASIP);
 		} else if (!strcasecmp(v->name, "logfile")) {
-         		ast_copy_string(gLogFile, v->value, sizeof(gLogFile));
+			if (v->value[0] == '/') {
+				ast_copy_string(gLogFile, v->value, sizeof(gLogFile));
+			} else {
+				snprintf(gLogFile, sizeof(gLogFile), "%s/%s", ast_config_AST_LOG_DIR, v->value);
+			}
 		} else if (!strcasecmp(v->name, "context")) {
          		ast_copy_string(gContext, v->value, sizeof(gContext));
          		ast_verb(3, "  == Setting default context to %s\n", gContext);
@@ -3838,9 +3843,9 @@ static int load_module(void)
 	if (!reload_config(0)) {
 
 		/* fire up the H.323 Endpoint */
-		if (OO_OK != ooH323EpInitialize(OO_CALLMODE_AUDIOCALL, gLogFile)) {
-         		ast_log(LOG_ERROR, "Failed to initialize OOH323 endpoint-"
-                            "OOH323 Disabled\n");
+		if (OO_OK != ooH323EpInitialize(OO_CALLMODE_AUDIOCALL, gLogFile, gInitError, sizeof(gInitError))) {
+			ast_log(LOG_ERROR, "Failed to initialize OOH323 endpoint: %s"
+				"OOH323 Disabled\n", gInitError);
 			ao2_ref(gCap, -1);
 			gCap = NULL;
 			ao2_ref(ooh323_tech.capabilities, -1);

addons/chan_ooh323.h

@@ -65,6 +65,7 @@
 #include "asterisk/udptl.h"
 #include "asterisk/stasis_channels.h"
 #include "asterisk/format_cache.h"
+#include "asterisk/paths.h"
 
 #include "ootypes.h"
 #include "ooUtils.h"

addons/ooh323c/README

@@ -75,7 +75,7 @@ To run the stack test application chansetup
 3. For running calling instance
 
         ./h323peer [--use-ip ip] -n <number of calls> -duration <call duration>
-                   -interval <inetrval between successive calls> destination
+                   -interval <interval between successive calls> destination
 
    where all times are in seconds. Interval of 0 means next call will be placed
    after current call finishes. "destination" is the dotted ip address of the

addons/ooh323c/src/decode.c

@@ -417,7 +417,7 @@ int decodeDynBitString (OOCTXT* pctxt, ASN1DynBitStr* pBitStr)
    ASN1OCTET* ptmp;
    int nbits, stat = ASN_OK;
 
-   /* If "fast copy" option is not set (ASN1FATSCOPY) or if constructed,
+   /* If "fast copy" option is not set (ASN1FASTCOPY) or if constructed,
     * copy the bit string value into a dynamic memory buffer;
     * otherwise, store the pointer to the value in the decode
     * buffer in the data pointer argument. */
@@ -737,6 +737,10 @@ static int decodeOctets
          nbits -= 8;
       }
 
+      if (nbits <= 0) {
+         return ASN_OK;
+      }
+
       /* Copy last partial byte */
 
       if (nbits >= rshift) {
@@ -752,7 +756,7 @@ static int decodeOctets
 
          pctxt->buffer.bitOffset = 8 - nbitsInLastOctet;
       }
-      else if (nbits > 0) {  /* nbits < rshift */
+      else {  /* nbits > 0 && nbits < rshift */
          pbuffer[i] =
             pctxt->buffer.data[pctxt->buffer.byteIndex] << lshift;
          pctxt->buffer.bitOffset = rshift - nbits;
@@ -832,8 +836,8 @@ int decodeOpenType
 
 int decodeSemiConsInteger (OOCTXT* pctxt, ASN1INT* pvalue, ASN1INT lower)
 {
-   signed char b;
-   unsigned char ub;
+   signed char b = 0;
+   unsigned char ub = 0;
    ASN1UINT nbytes;
    int stat;
 

addons/ooh323c/src/encode.c

@@ -1007,7 +1007,7 @@ int encode2sCompBinInt (OOCTXT* pctxt, ASN1INT value)
 {
    /* 10.4.6  A minimum octet 2's-complement-binary-integer encoding    */
    /* of the whole number has a field width that is a multiple of 8     */
-   /* bits and also satisifies the condition that the leading 9 bits    */
+   /* bits and also satisfies the condition that the leading 9 bits     */
    /* field shall not be all zeros and shall not be all ones.           */
 
    /* first encode integer value into a local buffer */
@@ -1048,7 +1048,7 @@ static int encodeNonNegBinInt (OOCTXT* pctxt, ASN1UINT value)
    /* 10.3.6  A minimum octet non-negative binary integer encoding of   */
    /* the whole number (which does not predetermine the number of       */
    /* octets to be used for the encoding) has a field which is a        */
-   /* multiple of 8 bits and also satisifies the condition that the     */
+   /* multiple of 8 bits and also satisfies the condition that the      */
    /* leading eight bits of the field shall not be zero unless the      */
    /* field is precisely 8 bits long.                                   */
 

addons/ooh323c/src/eventHandler.h

@@ -49,7 +49,7 @@ extern "C" {
  * SEQUENCE OF, SET OF, or CHOICE construct is parsed.
  *
  * @param name         For SEQUENCE, SET, or CHOICE, this is the name of the
- *                       element as defined in the ASN.1 defination. For
+ *                       element as defined in the ASN.1 definition. For
  *                       SEQUENCE OF or SET OF, this is set to the name
  *                       "element".
  * @param index        For SEQUENCE, SET, or CHOICE, this is not used and is
@@ -68,7 +68,7 @@ typedef void (*StartElement) (const char* name, int index) ;
  * SEQUENCE, SET, SEQUENCE OF, SET OF, or CHOICE construct.
  *
  * @param name         For SEQUENCE, SET, or CHOICE, this is the name of the
- *                       element as defined in the ASN.1 defination. For
+ *                       element as defined in the ASN.1 definition. For
  *                       SEQUENCE OF or SET OF, this is set to the name
  *                       "element".
  * @param index        For SEQUENCE, SET, or CHOICE, this is not used and is
@@ -194,7 +194,7 @@ typedef void (*EnumValue) (ASN1UINT value) ;
  * within a decode function when an ASN.1 open type is parsed.
  *
  * @param numocts      Number of octets in the parsed value.
- * @param data         Pointer to byet array contain in tencoded ASN.1
+ * @param data         Pointer to byte array contain in tencoded ASN.1
  *                       value.
  * @return             - none
  */

addons/ooh323c/src/h323/H323-MESSAGES.h

@@ -1926,7 +1926,7 @@ typedef struct EXTERN H225SecurityCapabilities {
    } m;
    H225NonStandardParameter nonStandard;
    H225SecurityServiceMode encryption;
-   H225SecurityServiceMode authenticaton;
+   H225SecurityServiceMode authentication;
    H225SecurityServiceMode integrity;
 } H225SecurityCapabilities;
 

addons/ooh323c/src/h323/H323-MESSAGESDec.c

@@ -6357,14 +6357,14 @@ EXTERN int asn1PD_H225SecurityCapabilities (OOCTXT* pctxt, H225SecurityCapabilit
 
    invokeEndElement (pctxt, "encryption", -1);
 
-   /* decode authenticaton */
+   /* decode authentication */
 
-   invokeStartElement (pctxt, "authenticaton", -1);
+   invokeStartElement (pctxt, "authentication", -1);
 
-   stat = asn1PD_H225SecurityServiceMode (pctxt, &pvalue->authenticaton);
+   stat = asn1PD_H225SecurityServiceMode (pctxt, &pvalue->authentication);
    if (stat != ASN_OK) return stat;
 
-   invokeEndElement (pctxt, "authenticaton", -1);
+   invokeEndElement (pctxt, "authentication", -1);
 
    /* decode integrity */
 

addons/ooh323c/src/h323/H323-MESSAGESEnc.c

@@ -4408,9 +4408,9 @@ EXTERN int asn1PE_H225SecurityCapabilities (OOCTXT* pctxt, H225SecurityCapabilit
    stat = asn1PE_H225SecurityServiceMode (pctxt, &pvalue->encryption);
    if (stat != ASN_OK) return stat;
 
-   /* encode authenticaton */
+   /* encode authentication */
 
-   stat = asn1PE_H225SecurityServiceMode (pctxt, &pvalue->authenticaton);
+   stat = asn1PE_H225SecurityServiceMode (pctxt, &pvalue->authentication);
    if (stat != ASN_OK) return stat;
 
    /* encode integrity */

addons/ooh323c/src/memheap.c

@@ -265,7 +265,7 @@ void* memHeapAlloc (void** ppvMemHeap, int nbytes)
                   if (nunits <= (ASN1UINT)pElem_nunits (pElem)) {
                      RTMEMDIAG3
                         ("memHeapAlloc: "
-                         "found an exisiting free element 0x%x, size %d\n",
+                         "found an existing free element 0x%x, size %d\n",
                         pElem, (pElem_nunits (pElem) * 8u));
 
                      if (pMemBlk->freeElemOff ==
@@ -373,7 +373,7 @@ void* memHeapAlloc (void** ppvMemHeap, int nbytes)
          CHECKMEMBLOCK (pMemHeap, pMemBlk);
       }
       else  {
-	 ast_mutex_unlock(&pMemHeap->pLock);
+	      ast_mutex_unlock(&pMemHeap->pLock);
          return NULL;
       }
    }
@@ -753,9 +753,10 @@ void* memHeapRealloc (void** ppvMemHeap, void* mem_p, int nbytes_)
              if (newMemBlk == 0)
                 return 0;
              pMemLink->pMemBlk = newMemBlk;
-	 }
-         else
+	      }
+         else {
             return 0;
+         }
          *(int*)(((char*)pMemLink) + sizeof (OSMemLink)) = nbytes_;
          return pMemLink->pMemBlk;
       }
@@ -1132,7 +1133,7 @@ void* memHeapMarkSaved (void** ppvMemHeap, const void* mem_p,
          RTMEMDIAG2 ("memHeapMarkSaved: the element 0x%x is "
                          "already free!\n", pElem);
 
-	 ast_mutex_unlock(&pMemHeap->pLock);
+	      ast_mutex_unlock(&pMemHeap->pLock);
          return 0;
       }
 
@@ -1151,9 +1152,10 @@ void* memHeapMarkSaved (void** ppvMemHeap, const void* mem_p,
             CLEAR_SAVED (pMemBlk, pElem);
          nsaved = pMemBlk->nsaved;
       }
-      else
-	 ast_mutex_unlock(&pMemHeap->pLock);
-         return 0;
+      else {
+	      ast_mutex_unlock(&pMemHeap->pLock);
+      }
+      return 0;
    }
    if (saved && nsaved > 0)
       pMemLink->blockType |= RTMEMSAVED;
@@ -1212,7 +1214,7 @@ static OSMemLink* memHeapAddBlock (OSMemLink** ppMemLink,
 
    /* if pMemBlk has RTMEMLINK flags it means that it is allocated
     * cooperatively with OSMemLink, and we don't need to do additional
-    * allocations for it. Just use pointer's arithemtic. */
+    * allocations for it. Just use pointer's arithmetic. */
 
    if (blockType & RTMEMLINK)
       pMemLink = (OSMemLink*) (((ASN1OCTET*)pMemBlk) - sizeof (OSMemLink));

addons/ooh323c/src/ooCalls.c

@@ -783,7 +783,7 @@ OOH323CallData* ooFindCallByToken(const char *callToken)
 
 
 
-/* Checks whether session with suplied ID and direction is already active*/
+/* Checks whether session with supplied ID and direction is already active*/
 ASN1BOOL ooIsSessionEstablished(OOH323CallData *call, int sessionID, char* dir)
 {
    OOLogicalChannel * temp = NULL;

addons/ooh323c/src/ooCapability.h

@@ -294,7 +294,7 @@ EXTERN int ooCapabilityDisableDTMFQ931Keypad(struct OOH323CallData *call);
  * and txframes parameters to the endpoint or call.(ex. G711, G728, G723.1,
  * G729)
  * @param call                 Handle to a call. If this is not Null, then
- *                             capability is added to call's remote enpoint
+ *                             capability is added to call's remote endpoint
  *                             capability list, else it is added to local H323
  *                             endpoint list.
  * @param cap                  Type of G711 capability to be added.
@@ -326,7 +326,7 @@ EXTERN int ooCapabilityAddSimpleCapability
  * to local endpoints capability list or to remote endpoints capability list or
  * to a call's capability list.
  * @param call                 Handle to a call. If this is not Null, then
- *                             capability is added to call's remote enpoint
+ *                             capability is added to call's remote endpoint
  *                             capability list, else it is added to local H323
  *                             endpoint list.
  * @param cap                  Type of GSM capability to be added.
@@ -357,7 +357,7 @@ int ooCapabilityAddGSMCapability(struct OOH323CallData *call, int cap,
  * capability list or to remote endpoints capability list or to a call's
  * capability list.
  * @param call                 Handle to a call. If this is not Null, then
- *                             capability is added to call's remote enpoint
+ *                             capability is added to call's remote endpoint
  *                             capability list, else it is added to local H323
  *                             endpoint list.
  * @param sqcifMPI             Minimum picture interval for encoding/decoding
@@ -396,7 +396,7 @@ EXTERN int ooCapabilityAddH263VideoCapability(struct OOH323CallData *call,
 /**
  * This function is an helper function to ooCapabilityAddH263VideoCapability.
  * @param call                 Handle to a call. If this is not Null, then
- *                             capability is added to call's remote enpoint
+ *                             capability is added to call's remote endpoint
  *                             capability list, else it is added to local H323
  *                             endpoint list.
  * @param sqcifMPI             Minimum picture interval for encoding/decoding
@@ -717,11 +717,11 @@ EXTERN int ooAppendCapToCapPrefs(struct OOH323CallData *call, int cap);
 EXTERN int ooChangeCapPrefOrder(struct OOH323CallData *call, int cap, int pos);
 
 /**
- * This function is used to preppend a particular capability to preference
+ * This function is used to prepend a particular capability to preference
  * list.
  * @param call     Handle to call, if call's preference list has to be modified
  *                 else NULL, to modify endpoint's preference list.
- * @param cap      Capability to be preppended.
+ * @param cap      Capability to be prepended.
  *
  * @return         OO_OK, on success. OO_FAILED, otherwise.
  */

addons/ooh323c/src/ooCmdChannel.c

@@ -186,7 +186,7 @@ int ooReadAndProcessStackCommand()
                 if(!pCall) {
                    OOTRACEINFO2("Call \"%s\" does not exist\n",
                                 (char*)cmd.param1);
-                   OOTRACEINFO1("Call migth be cleared/closed\n");
+                   OOTRACEINFO1("Call might be cleared/closed\n");
                 }
                 else {
                      ooSendProgress(ooFindCallByToken((char*)cmd.param1));
@@ -200,7 +200,7 @@ int ooReadAndProcessStackCommand()
                   if(!pCall) {
                      OOTRACEINFO2("Call \"%s\" does not exist\n",
                                   (char*)cmd.param1);
-                     OOTRACEINFO1("Call migth be cleared/closed\n");
+                     OOTRACEINFO1("Call might be cleared/closed\n");
                   }
                   else {
                      ooSendAlerting(ooFindCallByToken((char*)cmd.param1));

addons/ooh323c/src/ooGkClient.c

@@ -1231,7 +1231,7 @@ int ooGkClientHandleRegistrationConfirm
                                 (&pGkClient->ctxt, sizeof(ooGkClientTimerCb));
       if(!cbData)
       {
-         OOTRACEERR1("Error:Failed to allocate memory for Regisration timer."
+         OOTRACEERR1("Error:Failed to allocate memory for Registration timer."
                      "\n");
          pGkClient->state = GkClientFailed;
          return OO_FAILED;
@@ -1927,7 +1927,7 @@ int ooGkClientSendAdmissionRequest
                                (&pGkClient->ctxt, sizeof(ooGkClientTimerCb));
    if(!cbData)
    {
-      OOTRACEERR1("Error:Failed to allocate memory for Regisration timer."
+      OOTRACEERR1("Error:Failed to allocate memory for Registration timer."
                   "\n");
       pGkClient->state = GkClientFailed;
       ast_mutex_unlock(&pGkClient->Lock);

addons/ooh323c/src/ooGkClient.h

@@ -318,7 +318,7 @@ EXTERN int ooGkClientHandleRASMessage
 
 
 /**
- * This function is used to send a message on Gatekeeper clien't RAS channel.
+ * This function is used to send a message on Gatekeeper client RAS channel.
  * @param pGkClient   Handle to the gatekeeper client.
  * @param pRasMsg     Handle to Ras message to be sent.
  *

addons/ooh323c/src/ooLogChan.c

@@ -78,7 +78,7 @@ OOLogicalChannel* ooAddNewLogicalChannel(OOH323CallData *call, int channelNo,
       OOTRACEDBGC3("Using configured media info (%s, %s)\n", call->callType,
                    call->callToken);
       pNewChannel->localRtpPort = pMediaInfo->lMediaRedirPort ? pMediaInfo->lMediaRedirPort : pMediaInfo->lMediaPort;
-      /* check MediaRedirPort here because RedirCPort is ReditPort + 1 and can't be 0 ;) */
+      /* check MediaRedirPort here because RedirCPort is RedirPort + 1 and can't be 0 ;) */
       pNewChannel->localRtcpPort = pMediaInfo->lMediaRedirPort ? pMediaInfo->lMediaRedirCPort : pMediaInfo->lMediaCntrlPort;
       /* If user application has not specified a specific ip and is using
          multihomed mode, substitute appropriate ip.

addons/ooh323c/src/ooSocket.c

@@ -389,7 +389,7 @@ int ooSocketAccept (OOSOCKET socket, OOSOCKET *pNewSocket,
 
    if (destAddr != 0) {
       if ((host = ast_sockaddr_stringify_addr(&addr)) != NULL)
-      	strncpy(destAddr, host, strlen(host));
+       memcpy(destAddr, host, strlen(host) + 1);
    }
    if (destPort != 0)
       *destPort =  ast_sockaddr_port(&addr);
@@ -537,8 +537,6 @@ int ooPDWrite(struct pollfd *pfds, int nfds, int fd)
 int ooGetLocalIPAddress(char * pIPAddrs)
 {
    int ret;
-   struct hostent *hp;
-   struct ast_hostent phost;
    char hostname[100];
 
    if(pIPAddrs == NULL)
@@ -546,20 +544,11 @@ int ooGetLocalIPAddress(char * pIPAddrs)
    ret = gethostname(hostname, 100);
    if(ret == 0)
    {
-      if ((hp = ast_gethostbyname(hostname, &phost))) {
-			if (hp->h_addrtype == AF_INET6) {
-				struct in6_addr i;
-				memcpy(&i, hp->h_addr, sizeof(i));
-				strcpy(pIPAddrs, (inet_ntop(AF_INET6, &i,
-				hostname, sizeof(hostname))) == NULL ? "::1" :
-				inet_ntop(AF_INET6, &i, hostname, sizeof(hostname)));
-			} else {
-	  			struct in_addr i;
-				memcpy(&i, hp->h_addr, sizeof(i));
-			  	strcpy(pIPAddrs, (ast_inet_ntoa(i) == NULL) ? "127.0.0.1" : ast_inet_ntoa(i));
-			}
-      } else {
+      struct ast_sockaddr addr = { {0,} };
+      if (ast_sockaddr_resolve_first_af(&addr, hostname, PARSE_PORT_FORBID, AF_UNSPEC)) {
          return -1;
+      } else {
+         strcpy(pIPAddrs, ast_sockaddr_stringify_addr(&addr));
       }
    }
    else{

addons/ooh323c/src/ooSocket.h

@@ -318,7 +318,7 @@ EXTERN int ooSocketSendTo(OOSOCKET socket, const ASN1OCTET* pdata,
 
 /**
  * This function is used for synchronous monitoring of multiple sockets.
- * For more information refer to documnetation of "select" system call.
+ * For more information refer to documentation of "select" system call.
  *
  * @param nfds         The highest numbered descriptor to be monitored
  *                     plus one.

addons/ooh323c/src/ooUtils.c

@@ -30,7 +30,7 @@ OOBOOL ooUtilsIsStrEmpty (const char* str)
 }
 
 
-OOBOOL ooIsDailedDigit(const char* str)
+OOBOOL ooIsDialedDigit(const char* str)
 {
    if(str == NULL || *str =='\0') { return FALSE; }
    while(*str != '\0')

addons/ooh323c/src/ooUtils.h

@@ -48,6 +48,6 @@ EXTERN OOBOOL ooUtilsIsStrEmpty (const char * str);
  * @param str    String to test
  * @return       TRUE if string contains all digits; FALSE otherwise
  */
-EXTERN OOBOOL ooIsDailedDigit(const char* str);
+EXTERN OOBOOL ooIsDialedDigit(const char* str);
 
 #endif

addons/ooh323c/src/ooasn1.h

@@ -607,7 +607,7 @@ EXTERN int initContextBuffer
 /**
  * This function initializes a context block. It makes sure that if the block
  * was not previosly initialized, that all key working parameters are set to
- * thier correct initial state values (i.e. declared within a function as a
+ * their correct initial state values (i.e. declared within a function as a
  * normal working variable), it is required that they invoke this function
  * before using it.
  *
@@ -664,7 +664,7 @@ EXTERN int setPERBufferUsingCtxt (OOCTXT* pTarget, OOCTXT* pSource);
 /**
  * This function adds an integer parameter to an error information structure.
  * Parameter substitution is done in much the same way as it is done in C
- * printf statments. The base error message specification that goes along with
+ * printf statements. The base error message specification that goes along with
  * a particular status code may have variable fields built in using '%'
  * modifiers. These would be replaced with actual parameter data.
  *
@@ -680,7 +680,7 @@ EXTERN int errAddIntParm (ASN1ErrInfo* pErrInfo, int errParm);
 /**
  * This function adds an string parameter to an error information structure.
  * Parameter substitution is done in much the same way as it is done in C
- * printf statments. The base error message specification that goes along with
+ * printf statements. The base error message specification that goes along with
  * a particular status code may have variable fields built in using '%'
  * modifiers. These would be replaced with actual parameter data.
  *
@@ -697,7 +697,7 @@ EXTERN int errAddStrParm (ASN1ErrInfo* pErrInfo,
 /**
  * This function adds an unsigned integer parameter to an error information
  * structure. Parameter substitution is done in much the same way as it is done
- * in C printf statments. The base error message specification that goes along
+ * in C printf statements. The base error message specification that goes along
  * with a particular status code may have variable fields built in using '%'
  * modifiers. These would be replaced with actual parameter data.
  *

addons/ooh323c/src/oochannels.c

@@ -297,7 +297,7 @@ int ooCreateH225Connection(OOH323CallData *call)
       {
          call->pH225Channel->sock = channelSocket;
 
-         OOTRACEINFO3("H2250 transmiter channel creation - successful "
+         OOTRACEINFO3("H2250 transmitter channel creation - successful "
                       "(%s, %s)\n", call->callType, call->callToken);
 
          /* If multihomed, get ip from socket */
@@ -454,7 +454,7 @@ int ooAcceptH225Connection()
                  call->callToken);
 
    if (remoteIP[0]) {
-	strncpy(call->remoteIP, remoteIP, strlen(remoteIP));
+	memcpy(call->remoteIP, remoteIP, strlen(remoteIP) + 1);
    }
 
    ast_mutex_unlock(&call->Lock);

addons/ooh323c/src/ooh245.c

@@ -2559,7 +2559,7 @@ int ooOnReceivedRequestChannelClose(OOH323CallData *call,
    if (lChannel->state == OO_LOGICALCHAN_ESTABLISHED) {
      ret = ooSendCloseLogicalChannel(call, lChannel);
      if (ret != OO_OK) {
-      OOTRACEERR3("ERROR:Failed to build CloseLgicalChannel message(%s, %s)\n",
+      OOTRACEERR3("ERROR:Failed to build CloseLogicalChannel message(%s, %s)\n",
                    call->callType, call->callToken);
       return OO_FAILED;
      }

addons/ooh323c/src/ooh245.h

@@ -174,7 +174,7 @@ EXTERN int ooSendMasterSlaveDeterminationReject (struct OOH323CallData* call);
  * MasterSlaveDetermination procedure.
  * @param call        Handle to the call for which MasterSlaveReject is
  *                    received.
- * @param reject      Poinetr to the received reject message.
+ * @param reject      Pointer to the received reject message.
  *
  * @return            OO_OK, on success. OO_FAILED, on failure.
  */
@@ -271,7 +271,7 @@ EXTERN int ooOnReceivedUserInputIndication
    (OOH323CallData *call, H245UserInputIndication *indication);
 
 /**
- * This function is called on receiving a TreminalCapabilitySetAck message.
+ * This function is called on receiving a TerminalCapabilitySetAck message.
  * If the MasterSlaveDetermination process is also over, this function
  * initiates the process of opening logical channels.
  * @param call          Pointer to call for which TCSAck is received.

addons/ooh323c/src/ooh323.c

@@ -616,7 +616,7 @@ int ooOnReceivedSetup(OOH323CallData *call, Q931Message *q931Msg)
               removeEventHandler(call->pctxt);
               return OO_FAILED;
          }
-         /* For now, just add decoded fast start elemts to list. This list
+         /* For now, just add decoded fast start elements to list. This list
             will be processed at the time of sending CONNECT message. */
          dListAppend(call->pctxt, &call->remoteFastStartOLCs, olc);
       }
@@ -1062,16 +1062,17 @@ int ooOnReceivedAlerting(OOH323CallData *call, Q931Message *q931Msg)
    if(q931Msg->userInfo->h323_uu_pdu.m.h245TunnelingPresent &&
       q931Msg->userInfo->h323_uu_pdu.h245Tunneling &&
 	 OO_TESTFLAG (call->flags, OO_M_TUNNELING)) {
-      if (alerting->m.h245AddressPresent)
+      if (alerting->m.h245AddressPresent) {
       	OOTRACEINFO3("Tunneling and h245address provided."
                      "Giving preference to Tunneling (%s, %s)\n",
                    	call->callType, call->callToken);
-	if (call->h225version >= 4) {
-		ret =ooSendTCSandMSD(call);
-	}
-	if (ret != OO_OK)
-		return ret;
-
+      }
+	   if (call->h225version >= 4) {
+		   ret =ooSendTCSandMSD(call);
+	   }
+	   if (ret != OO_OK) {
+		   return ret;
+      }
    } else if(alerting->m.h245AddressPresent) {
       if (OO_TESTFLAG (call->flags, OO_M_TUNNELING))
       {
@@ -1314,15 +1315,17 @@ int ooOnReceivedProgress(OOH323CallData *call, Q931Message *q931Msg)
    if(q931Msg->userInfo->h323_uu_pdu.m.h245TunnelingPresent &&
       q931Msg->userInfo->h323_uu_pdu.h245Tunneling &&
       OO_TESTFLAG (call->flags, OO_M_TUNNELING)) {
-      if (progress->m.h245AddressPresent)
+      if (progress->m.h245AddressPresent) {
       	OOTRACEINFO3("Tunneling and h245address provided."
                      "Giving preference to Tunneling (%s, %s)\n",
                      call->callType, call->callToken);
-	if (call->h225version >= 4) {
-		ret =ooSendTCSandMSD(call);
-	}
-	if (ret != OO_OK)
+      }
+	   if (call->h225version >= 4) {
+		   ret =ooSendTCSandMSD(call);
+	   }
+	   if (ret != OO_OK) {
 		return ret;
+      }
    } else if(progress->m.h245AddressPresent) {
       if (OO_TESTFLAG (call->flags, OO_M_TUNNELING))
       {

addons/ooh323c/src/ooh323ep.c

@@ -34,7 +34,7 @@ ast_mutex_t bindPortLock;
 extern DList g_TimerList;
 
 int ooH323EpInitialize
-   (enum OOCallMode callMode, const char* tracefile)
+   (enum OOCallMode callMode, const char* tracefile, char* errstr, int errstr_max)
 {
 
    memset(&gH323ep, 0, sizeof(ooEndPoint));
@@ -46,7 +46,7 @@ int ooH323EpInitialize
    {
       if(strlen(tracefile)>= MAXFILENAME)
       {
-         printf("Error:File name longer than allowed maximum %d\n",
+         snprintf(errstr, errstr_max, "Error:File name longer than allowed maximum %d\n",
                  MAXFILENAME-1);
          return OO_FAILED;
       }
@@ -59,13 +59,13 @@ int ooH323EpInitialize
    gH323ep.fptraceFile = fopen(gH323ep.traceFile, "a");
    if(gH323ep.fptraceFile == NULL)
    {
-      printf("Error:Failed to open trace file %s for write.\n",
+      snprintf(errstr, errstr_max, "Error:Failed to open trace file %s for write.\n",
                   gH323ep.traceFile);
       return OO_FAILED;
    }
 
    /* Initialize default port ranges that will be used by stack.
-      Apps can override these by explicitely setting port ranges
+      Apps can override these by explicitly setting port ranges
    */
 
    gH323ep.tcpPorts.start = TCPPORTSSTART;

addons/ooh323c/src/ooh323ep.h

@@ -163,7 +163,7 @@ typedef struct OOH323EndPoint {
  * @return               OO_OK, on success. OO_FAILED, on failure
  */
 EXTERN int ooH323EpInitialize
-   (enum OOCallMode callMode, const char* tracefile);
+   (enum OOCallMode callMode, const char* tracefile, char* errstr, int errstr_max);
 
 /**
  * This function is used to represent the H.323 application endpoint as

addons/ooh323c/src/ooq931.c

@@ -192,11 +192,13 @@ EXTERN int ooQ931Decode
          screening indicators ;-) */
       if(ie->discriminator == Q931CallingPartyNumberIE)
       {
+         int numoffset=1;
          OOTRACEDBGB1("   CallingPartyNumber IE = {\n");
-         if(ie->length < OO_MAX_NUMBER_LENGTH)
+         if(!(0x80 & ie->data[0])) numoffset = 2;
+
+         if( (ie->length >= numoffset) &&
+             (ie->length < OO_MAX_NUMBER_LENGTH) )
          {
-            int numoffset=1;
-            if(!(0x80 & ie->data[0])) numoffset = 2;
             memcpy(number, ie->data+numoffset,ie->length-numoffset);
             number[ie->length-numoffset]='\0';
             OOTRACEDBGB2("      %s\n", number);
@@ -204,7 +206,7 @@ EXTERN int ooQ931Decode
                ooCallSetCallingPartyNumber(call, number);
          }
          else{
-            OOTRACEERR3("Error:Calling party number too long. (%s, %s)\n",
+            OOTRACEERR3("Error:Calling party number outside range. (%s, %s)\n",
                            call->callType, call->callToken);
          }
          OOTRACEDBGB1("   }\n");
@@ -214,7 +216,8 @@ EXTERN int ooQ931Decode
       if(ie->discriminator == Q931CalledPartyNumberIE)
       {
          OOTRACEDBGB1("   CalledPartyNumber IE = {\n");
-         if(ie->length < OO_MAX_NUMBER_LENGTH)
+         if( (ie->length >= 1) &&
+             (ie->length < OO_MAX_NUMBER_LENGTH) )
          {
             memcpy(number, ie->data+1,ie->length-1);
             number[ie->length-1]='\0';
@@ -223,7 +226,7 @@ EXTERN int ooQ931Decode
                ooCallSetCalledPartyNumber(call, number);
          }
          else{
-            OOTRACEERR3("Error:Calling party number too long. (%s, %s)\n",
+            OOTRACEERR3("Error:Calling party number outside range. (%s, %s)\n",
                            call->callType, call->callToken);
          }
          OOTRACEDBGB1("   }\n");
@@ -585,7 +588,7 @@ int ooDecodeUUIE(OOCTXT* pctxt, Q931Message *q931Msg)
    unsigned int i;
    ASN1BOOL aligned=TRUE;
    int stat;
-   Q931InformationElement *ie;
+   Q931InformationElement *ie=NULL;
    /* OOCTXT *pctxt = &gH323ep.msgctxt; */
    if(q931Msg ==NULL)
    {
@@ -1210,7 +1213,7 @@ int ooSetFastStartResponse(OOH323CallData *pCall, Q931Message *pQ931msg,
          pChannel = ooFindLogicalChannelByLogicalChannelNo
                       (pCall, olc->forwardLogicalChannelNumber);
 
-         /* start receive and tramsmit channel listening */
+         /* start receive and transmit channel listening */
          if(dir & OORX)
          {
             strcpy(pChannel->remoteIP, remoteMediaControlIP);
@@ -2025,7 +2028,7 @@ int ooSendStatusInquiry(OOH323CallData *call)
    /* OOCTXT *pctxt = &gH323ep.msgctxt; */
    OOCTXT *pctxt = call->msgctxt;
 
-   OOTRACEDBGC3("Building StatusInquryMsg (%s, %s)\n", call->callType,
+   OOTRACEDBGC3("Building StatusInquiryMsg (%s, %s)\n", call->callType,
                  call->callToken);
    ret = ooCreateQ931Message(pctxt, &q931msg, Q931StatusEnquiryMsg);
    if(ret != OO_OK)
@@ -2987,7 +2990,7 @@ int ooH323MakeCall_helper(OOH323CallData *call)
          }
          if(!epCap)
          {
-            OOTRACEWARN4("Warn:Preferred capability %s is abscent in "
+            OOTRACEWARN4("Warn:Preferred capability %s is absent in "
                          "capability list. (%s, %s)\n",
                          ooGetCapTypeText(call->capPrefs.order[k]),
                          call->callType, call->callToken);

addons/ooh323c/src/ooq931.h

@@ -582,7 +582,7 @@ EXTERN int ooH323MakeCall_helper(struct OOH323CallData *call);
  * @param dest      Destination string to be parsed.
  * @param parsedIP  Pointer to buffer in which parsed ip:port will be returned.
  * @param len       Length of the buffer passed.
- * @param aliasList Aliase List in which new aliases will be added.
+ * @param aliasList Aliases List in which new aliases will be added.
  *
  * @return          OO_OK, on success. OO_FAILED, on failure.
  */

addons/res_config_mysql.c

@@ -554,7 +554,7 @@ static struct ast_config *realtime_multi_mysql(const char *database, const char
 static int update_mysql(const char *database, const char *tablename, const char *keyfield, const char *lookup, const struct ast_variable *rt_fields)
 {
 	struct mysql_conn *dbh;
-	my_ulonglong numrows;
+	uint64_t numrows;
 	const struct ast_variable *field = rt_fields;
 	struct ast_str *sql = ast_str_thread_get(&sql_buf, 100), *buf = ast_str_thread_get(&scratch_buf, 100);
 	struct tables *table;
@@ -641,7 +641,7 @@ static int update_mysql(const char *database, const char *tablename, const char
 	release_table(table);
 	release_database(dbh);
 
-	ast_debug(1, "MySQL RealTime: Updated %llu rows on table: %s\n", numrows, tablename);
+	ast_debug(1, "MySQL RealTime: Updated %" PRIu64 " rows on table: %s\n", numrows, tablename);
 
 	/* From http://dev.mysql.com/doc/mysql/en/mysql-affected-rows.html
 	 * An integer greater than zero indicates the number of rows affected
@@ -655,7 +655,7 @@ static int update_mysql(const char *database, const char *tablename, const char
 static int update2_mysql(const char *database, const char *tablename, const struct ast_variable *lookup_fields, const struct ast_variable *update_fields)
 {
 	struct mysql_conn *dbh;
-	my_ulonglong numrows;
+	uint64_t numrows;
 	int first;
 	const struct ast_variable *field;
 	struct ast_str *sql = ast_str_thread_get(&sql_buf, 100), *buf = ast_str_thread_get(&scratch_buf, 100);
@@ -738,7 +738,7 @@ static int update2_mysql(const char *database, const char *tablename, const stru
 	numrows = mysql_affected_rows(&dbh->handle);
 	release_database(dbh);
 
-	ast_debug(1, "MySQL RealTime: Updated %llu rows on table: %s\n", numrows, tablename);
+	ast_debug(1, "MySQL RealTime: Updated %" PRIu64 " rows on table: %s\n", numrows, tablename);
 
 	/* From http://dev.mysql.com/doc/mysql/en/mysql-affected-rows.html
 	 * An integer greater than zero indicates the number of rows affected
@@ -810,7 +810,7 @@ static int store_mysql(const char *database, const char *table, const struct ast
 static int destroy_mysql(const char *database, const char *table, const char *keyfield, const char *lookup, const struct ast_variable *rt_fields)
 {
 	struct mysql_conn *dbh;
-	my_ulonglong numrows;
+	uint64_t numrows;
 	struct ast_str *sql = ast_str_thread_get(&sql_buf, 16);
 	struct ast_str *buf = ast_str_thread_get(&scratch_buf, 16);
 	const struct ast_variable *field;
@@ -862,7 +862,7 @@ static int destroy_mysql(const char *database, const char *table, const char *ke
 	numrows = mysql_affected_rows(&dbh->handle);
 	release_database(dbh);
 
-	ast_debug(1, "MySQL RealTime: Deleted %llu rows on table: %s\n", numrows, table);
+	ast_debug(1, "MySQL RealTime: Deleted %" PRIu64 " rows on table: %s\n", numrows, table);
 
 	/* From http://dev.mysql.com/doc/mysql/en/mysql-affected-rows.html
 	 * An integer greater than zero indicates the number of rows affected
@@ -878,7 +878,7 @@ static struct ast_config *config_mysql(const char *database, const char *table,
 	struct mysql_conn *dbh;
 	MYSQL_RES *result;
 	MYSQL_ROW row;
-	my_ulonglong num_rows;
+	uint64_t num_rows;
 	struct ast_variable *new_v;
 	struct ast_category *cur_cat = NULL;
 	struct ast_str *sql = ast_str_thread_get(&sql_buf, 200);
@@ -916,7 +916,7 @@ static struct ast_config *config_mysql(const char *database, const char *table,
 
 	if ((result = mysql_store_result(&dbh->handle))) {
 		num_rows = mysql_num_rows(result);
-		ast_debug(1, "MySQL RealTime: Found %llu rows.\n", num_rows);
+		ast_debug(1, "MySQL RealTime: Found %" PRIu64 " rows.\n", num_rows);
 
 		/* There might exist a better way to access the column names other than counting,
 		 * but I believe that would require another loop that we don't need. */

agi/eagi-test.c

@@ -93,7 +93,7 @@ static char *wait_result(void)
 				fprintf(stderr, "Got %d/%d bytes of audio\n", res, bytes);
 #endif
 				bytes += res;
-				/* Prentend we detected some audio after 3 seconds */
+				/* Pretend we detected some audio after 3 seconds */
 				if (bytes > 16000 * 3) {
 					return "Sample Message";
 					bytes = 0;

agi/jukebox.agi

@@ -348,7 +348,7 @@ sub music_dir_cache {
 }
 
 # this function will fill the @masterCacheList of all the files that
-# need to have text2speeced ulaw files of their names generated
+# need to have text2speech ulaw files of their names generated
 sub music_dir_cache_genlist {
 	my $dir = shift;
 	if (!opendir(THEDIR, rmts($MUSIC.$dir))) {

apps/app_adsiprog.c

@@ -588,7 +588,7 @@ static struct adsi_subscript *getsubbyname(struct adsi_script *state, char *name
 	}
 
 	if (state->numsubs > 127) {
-		ast_log(LOG_WARNING, "No more subscript space at line %d of %s\n", lineno, script);
+		ast_log(LOG_WARNING, "No more subscript space at line %d of %s\n", lineno, S_OR(script, "unknown"));
 		return NULL;
 	}
 

apps/app_agent_pool.c

@@ -24,8 +24,14 @@
  *
  * See Also:
  * \arg \ref AstCREDITS
- * \arg \ref Config_agent
+ * \arg \ref agents.conf "Config_agent"
  */
+
+/*!
+ * \page agents.conf agents.conf
+ * \verbinclude agents.conf.sample
+ */
+
 /*** MODULEINFO
 	<support_level>core</support_level>
  ***/
@@ -92,8 +98,8 @@
 			<ref type="application">PauseQueueMember</ref>
 			<ref type="application">UnpauseQueueMember</ref>
 			<ref type="function">AGENT</ref>
-			<ref type="function">CHANNEL(dtmf_features)</ref>
-			<ref type="function">CONNECTEDLINE()</ref>
+			<ref type="function">CHANNEL</ref>
+			<ref type="function">CONNECTEDLINE</ref>
 			<ref type="filename">agents.conf</ref>
 			<ref type="filename">queues.conf</ref>
 		</see-also>
@@ -456,17 +462,11 @@ struct agents_cfg {
 	struct ao2_container *agents;
 };
 
-static const char *agent_type_blacklist[] = {
-	"general",
-	"agents",
-	NULL,
-};
-
 static struct aco_type agent_type = {
 	.type = ACO_ITEM,
 	.name = "agent-id",
-	.category_match = ACO_BLACKLIST_ARRAY,
-	.category = (const char *)agent_type_blacklist,
+	.category_match = ACO_BLACKLIST_EXACT,
+	.category = "general",
 	.item_alloc = agent_cfg_alloc,
 	.item_find = agent_cfg_find,
 	.item_offset = offsetof(struct agents_cfg, agents),
@@ -652,8 +652,6 @@ static struct ao2_container *agents;
  * \brief Lock the agent.
  *
  * \param agent Agent to lock
- *
- * \return Nothing
  */
 #define agent_lock(agent)	_agent_lock(agent, __FILE__, __PRETTY_FUNCTION__, __LINE__, #agent)
 static inline void _agent_lock(struct agent_pvt *agent, const char *file, const char *function, int line, const char *var)
@@ -665,8 +663,6 @@ static inline void _agent_lock(struct agent_pvt *agent, const char *file, const
  * \brief Unlock the agent.
  *
  * \param agent Agent to unlock
- *
- * \return Nothing
  */
 #define agent_unlock(agent)	_agent_unlock(agent, __FILE__, __PRETTY_FUNCTION__, __LINE__, #agent)
 static inline void _agent_unlock(struct agent_pvt *agent, const char *file, const char *function, int line, const char *var)
@@ -684,8 +680,6 @@ static inline void _agent_unlock(struct agent_pvt *agent, const char *file, cons
  * \note Assumes the agent lock is already obtained.
  *
  * \note Defined locking order is channel lock then agent lock.
- *
- * \return Nothing
  */
 static struct ast_channel *agent_lock_logged(struct agent_pvt *agent)
 {
@@ -750,8 +744,6 @@ static enum ast_device_state agent_pvt_devstate_get(const char *agent_id)
  * \since 12.0.0
  *
  * \param agent_id Which agent needs the device state updated.
- *
- * \return Nothing
  */
 static void agent_devstate_changed(const char *agent_id)
 {
@@ -1011,8 +1003,6 @@ AST_MUTEX_DEFINE_STATIC(agent_holding_lock);
  * \param payload_size Size of the payload if payload is non-NULL.  A number otherwise.
  *
  * \note The payload MUST NOT have any resources that need to be freed.
- *
- * \return Nothing
  */
 static void clear_agent_status(struct ast_bridge_channel *bridge_channel, const void *payload, size_t payload_size)
 {
@@ -1028,8 +1018,6 @@ static void clear_agent_status(struct ast_bridge_channel *bridge_channel, const
  * \param agent Which agent is connecting to the caller.
  *
  * \note The agent is locked on entry and not locked on exit.
- *
- * \return Nothing
  */
 static void agent_connect_caller(struct ast_bridge_channel *bridge_channel, struct agent_pvt *agent)
 {
@@ -1363,8 +1351,6 @@ static int bridge_agent_hold_push(struct ast_bridge *self, struct ast_bridge_cha
  * bridge_channel->bridge_pvt.
  *
  * \note On entry, self is already locked.
- *
- * \return Nothing
  */
 static void bridge_agent_hold_pull(struct ast_bridge *self, struct ast_bridge_channel *bridge_channel)
 {
@@ -1382,8 +1368,6 @@ static void bridge_agent_hold_pull(struct ast_bridge *self, struct ast_bridge_ch
  * references to the bridge so it can be destroyed.
  *
  * \note On entry, self must NOT be locked.
- *
- * \return Nothing
  */
 static void bridge_agent_hold_dissolving(struct ast_bridge *self)
 {
@@ -1457,9 +1441,9 @@ static void send_agent_logoff(struct ast_channel *chan, const char *agent, long
 
 	ast_assert(agent != NULL);
 
-	blob = ast_json_pack("{s: s, s: i}",
+	blob = ast_json_pack("{s: s, s: I}",
 		"agent", agent,
-		"logintime", logintime);
+		"logintime", (ast_json_int_t)logintime);
 	if (!blob) {
 		return;
 	}
@@ -1475,8 +1459,6 @@ static void send_agent_logoff(struct ast_channel *chan, const char *agent, long
  * \param agent Which agent logging out.
  *
  * \note On entry agent is already locked.  On exit it is no longer locked.
- *
- * \return Nothing
  */
 static void agent_logout(struct agent_pvt *agent)
 {
@@ -1514,8 +1496,6 @@ static void agent_logout(struct agent_pvt *agent)
  *
  * \param agent Which agent.
  * \param logged The logged in channel.
- *
- * \return Nothing
  */
 static void agent_run(struct agent_pvt *agent, struct ast_channel *logged)
 {
@@ -2034,8 +2014,6 @@ static int agent_request_exec(struct ast_channel *chan, const char *data)
  *
  * \param agent What to setup channel config values on.
  * \param chan Channel logging in as an agent.
- *
- * \return Nothing
  */
 static void agent_login_channel_config(struct agent_pvt *agent, struct ast_channel *chan)
 {

apps/app_alarmreceiver.c

@@ -158,7 +158,7 @@ static const char app[] = "AlarmReceiver";
 			events to the standard input of the application.
 			The configuration file also contains settings for DTMF timing, and for the loudness of the
 			acknowledgement tones.</para>
-			<note><para>Few Ademco DTMF signalling formats are detected automaticaly: Contact ID, Express 4+1,
+			<note><para>Few Ademco DTMF signalling formats are detected automatically: Contact ID, Express 4+1,
 			Express 4+2, High Speed and Super Fast.</para></note>
 			<para>The application is affected by the following variables:</para>
 			<variablelist>
@@ -203,7 +203,6 @@ static char event_file[14] = "/event-XXXXXX";
  * family, then create it and set its value to 1.
  *
  * \param key A database key to increment
- * \return Nothing
  */
 static void database_increment(char *key)
 {

apps/app_amd.c

@@ -92,6 +92,12 @@
 				<para>Is the maximum duration of a word to accept.</para>
 				<para>If exceeded, then the result is detection as a MACHINE</para>
 			</parameter>
+			<parameter name="audioFile" required="false">
+				<para>Is an audio file to play to the caller while AMD is in progress.</para>
+				<para>By default, no audio file is played.</para>
+				<para>If an audio file is configured in amd.conf, then that file will be used
+				if one is not specified here. That file may be overridden by this argument.</para>
+			</parameter>
 		</syntax>
 		<description>
 			<para>This application attempts to detect answering machines at the beginning
@@ -155,6 +161,9 @@ static int dfltBetweenWordsSilence  = 50;
 static int dfltMaximumNumberOfWords = 2;
 static int dfltSilenceThreshold     = 256;
 static int dfltMaximumWordLength    = 5000; /* Setting this to a large default so it is not used unless specify it in the configs or command line */
+static char *dfltAudioFile = NULL;
+
+static ast_mutex_t config_lock;
 
 /* Set to the lowest ms value provided in amd.conf or application parameters */
 static int dfltMaxWaitTimeForFrame  = 50;
@@ -162,8 +171,10 @@ static int dfltMaxWaitTimeForFrame  = 50;
 static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 {
 	int res = 0;
+	int audioFrameCount = 0;
 	struct ast_frame *f = NULL;
 	struct ast_dsp *silenceDetector = NULL;
+	struct timeval amd_tvstart;
 	int dspsilence = 0, framelength = 0;
 	RAII_VAR(struct ast_format *, readFormat, NULL, ao2_cleanup);
 	int inInitialSilence = 1;
@@ -177,7 +188,7 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 	char amdCause[256] = "", amdStatus[256] = "";
 	char *parse = ast_strdupa(data);
 
-	/* Lets set the initial values of the variables that will control the algorithm.
+	/* Let's set the initial values of the variables that will control the algorithm.
 	   The initial values are the default ones. If they are passed as arguments
 	   when invoking the application, then the default values will be overwritten
 	   by the ones passed as parameters. */
@@ -191,6 +202,7 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 	int silenceThreshold     = dfltSilenceThreshold;
 	int maximumWordLength    = dfltMaximumWordLength;
 	int maxWaitTimeForFrame  = dfltMaxWaitTimeForFrame;
+	const char *audioFile = NULL;
 
 	AST_DECLARE_APP_ARGS(args,
 		AST_APP_ARG(argInitialSilence);
@@ -202,8 +214,15 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 		AST_APP_ARG(argMaximumNumberOfWords);
 		AST_APP_ARG(argSilenceThreshold);
 		AST_APP_ARG(argMaximumWordLength);
+		AST_APP_ARG(audioFile);
 	);
 
+	ast_mutex_lock(&config_lock);
+	if (!ast_strlen_zero(dfltAudioFile)) {
+		audioFile = ast_strdupa(dfltAudioFile);
+	}
+	ast_mutex_unlock(&config_lock);
+
 	ast_verb(3, "AMD: %s %s %s (Fmt: %s)\n", ast_channel_name(chan),
 		S_COR(ast_channel_caller(chan)->ani.number.valid, ast_channel_caller(chan)->ani.number.str, "(N/A)"),
 		S_COR(ast_channel_redirecting(chan)->from.number.valid, ast_channel_redirecting(chan)->from.number.str, "(N/A)"),
@@ -231,6 +250,9 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 			silenceThreshold = atoi(args.argSilenceThreshold);
 		if (!ast_strlen_zero(args.argMaximumWordLength))
 			maximumWordLength = atoi(args.argMaximumWordLength);
+		if (!ast_strlen_zero(args.audioFile)) {
+			audioFile = args.audioFile;
+		}
 	} else {
 		ast_debug(1, "AMD using the default parameters.\n");
 	}
@@ -275,12 +297,20 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 	/* Set silence threshold to specified value */
 	ast_dsp_set_threshold(silenceDetector, silenceThreshold);
 
+	/* Set our start time so we can tie the loop to real world time and not RTP updates */
+	amd_tvstart = ast_tvnow();
+
+	/* Optional audio file to play to caller while AMD is doing its thing. */
+	if (!ast_strlen_zero(audioFile)) {
+		ast_streamfile(chan, audioFile, ast_channel_language(chan));
+	}
+
 	/* Now we go into a loop waiting for frames from the channel */
 	while ((res = ast_waitfor(chan, 2 * maxWaitTimeForFrame)) > -1) {
 		int ms = 0;
 
 		/* Figure out how long we waited */
-		if (res > 0) {
+		if (res >= 0) {
 			ms = 2 * maxWaitTimeForFrame - res;
 		}
 
@@ -293,7 +323,24 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 			break;
 		}
 
+		/* Check to make sure we haven't gone over our real-world timeout in case frames get stalled for whatever reason */
+		if ( (ast_tvdiff_ms(ast_tvnow(), amd_tvstart)) > totalAnalysisTime ) {
+			ast_frfree(f);
+			strcpy(amdStatus , "NOTSURE");
+			if ( audioFrameCount == 0 ) {
+				ast_verb(3, "AMD: Channel [%s]. No audio data received in [%d] seconds.\n", ast_channel_name(chan), totalAnalysisTime);
+				sprintf(amdCause , "NOAUDIODATA-%d", iTotalTime);
+				break;
+			}
+			ast_verb(3, "AMD: Channel [%s]. Timeout...\n", ast_channel_name(chan));
+			sprintf(amdCause , "TOOLONG-%d", iTotalTime);
+			break;
+		}
+
 		if (f->frametype == AST_FRAME_VOICE || f->frametype == AST_FRAME_CNG) {
+			/* keep track of the number of audio frames we get */
+			audioFrameCount++;
+
 			/* Figure out how long the frame is in milliseconds */
 			if (f->frametype == AST_FRAME_VOICE) {
 				framelength = (ast_codec_samples_count(f) / DEFAULT_SAMPLES_PER_MS);
@@ -410,6 +457,14 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 				}
 
 			}
+		} else {
+			iTotalTime += ms;
+			if (iTotalTime >= totalAnalysisTime) {
+				ast_frfree(f);
+				strcpy(amdStatus , "NOTSURE");
+				sprintf(amdCause , "TOOLONG-%d", iTotalTime);
+				break;
+			}
 		}
 		ast_frfree(f);
 	}
@@ -432,10 +487,14 @@ static void isAnsweringMachine(struct ast_channel *chan, const char *data)
 	/* Free the DSP used to detect silence */
 	ast_dsp_free(silenceDetector);
 
+	/* If we were playing something to pass the time, stop it now. */
+	if (!ast_strlen_zero(audioFile)) {
+		ast_stopstream(chan);
+	}
+
 	return;
 }
 
-
 static int amd_exec(struct ast_channel *chan, const char *data)
 {
 	isAnsweringMachine(chan, data);
@@ -486,7 +545,16 @@ static int load_config(int reload)
 					dfltMaximumNumberOfWords = atoi(var->value);
 				} else if (!strcasecmp(var->name, "maximum_word_length")) {
 					dfltMaximumWordLength = atoi(var->value);
-
+				} else if (!strcasecmp(var->name, "playback_file")) {
+					ast_mutex_lock(&config_lock);
+					if (dfltAudioFile) {
+						ast_free(dfltAudioFile);
+						dfltAudioFile = NULL;
+					}
+					if (!ast_strlen_zero(var->value)) {
+						dfltAudioFile = ast_strdup(var->value);
+					}
+					ast_mutex_unlock(&config_lock);
 				} else {
 					ast_log(LOG_WARNING, "%s: Cat:%s. Unknown keyword %s at line %d of amd.conf\n",
 						app, cat, var->name, var->lineno);
@@ -509,6 +577,12 @@ static int load_config(int reload)
 
 static int unload_module(void)
 {
+	ast_mutex_lock(&config_lock);
+	if (dfltAudioFile) {
+		ast_free(dfltAudioFile);
+	}
+	ast_mutex_unlock(&config_lock);
+	ast_mutex_destroy(&config_lock);
 	return ast_unregister_application(app);
 }
 
@@ -524,6 +598,7 @@ static int unload_module(void)
  */
 static int load_module(void)
 {
+	ast_mutex_init(&config_lock);
 	if (load_config(0) || ast_register_application_xml(app, amd_exec)) {
 		return AST_MODULE_LOAD_DECLINE;
 	}

apps/app_attended_transfer.c

@@ -0,0 +1,143 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2019, Alexei Gradinari
+ *
+ * Alexei Gradinari <alex2grad@gmail.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Attended transfer by caller channel
+ *
+ * \author Alexei Gradinari <alex2grad@gmail.com>
+ *
+ * \ingroup applications
+ */
+
+/*** MODULEINFO
+	<support_level>extended</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/app.h"
+#include "asterisk/channel.h"
+#include "asterisk/bridge.h"
+#include "asterisk/features_config.h"
+
+/*** DOCUMENTATION
+	<application name="AttendedTransfer" language="en_US">
+		<synopsis>
+			Attended transfer to the extension provided and TRANSFER_CONTEXT
+		</synopsis>
+		<syntax>
+			<parameter name="exten" required="true">
+				<para>Specify extension.</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Queue up attended transfer to the specified extension in the <literal>TRANSFER_CONTEXT</literal>.</para>
+			<para>Note that the attended transfer only work when two channels have answered and are bridged together.</para>
+			<para>Make sure to set Attended Transfer DTMF feature <literal>atxfer</literal>
+			and attended transfer is permitted.</para>
+			<para>The result of the application will be reported in the <variable>ATTENDEDTRANSFERSTATUS</variable>
+			channel variable:</para>
+			<variablelist>
+				<variable name="ATTENDEDTRANSFERSTATUS">
+					<value name="SUCCESS">
+						Transfer successfully queued.
+					</value>
+					<value name="FAILURE">
+						Transfer failed.
+					</value>
+					<value name="NOTPERMITTED">
+						Transfer not permitted.
+					</value>
+				</variable>
+			</variablelist>
+		</description>
+	</application>
+ ***/
+
+static const char * const app = "AttendedTransfer";
+
+static int attended_transfer_exec(struct ast_channel *chan, const char *data)
+{
+	char *exten = NULL;
+	const char *context = NULL;
+	char *parse;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(exten);
+	);
+	char feature_code[AST_FEATURE_MAX_LEN];
+	const char *digit;
+	struct ast_frame f = { .frametype = AST_FRAME_DTMF };
+
+	if (ast_strlen_zero((char *)data)) {
+		ast_log(LOG_WARNING, "%s requires an argument (exten)\n", app);
+		pbx_builtin_setvar_helper(chan, "ATTENDEDTRANSFERSTATUS", "FAILURE");
+		return 0;
+	}
+
+	context = pbx_builtin_getvar_helper(chan, "TRANSFER_CONTEXT");
+	if (ast_strlen_zero(context)) {
+		pbx_builtin_setvar_helper(chan, "ATTENDEDTRANSFERSTATUS", "NOTPERMITTED");
+		return 0;
+	}
+
+	ast_channel_lock(chan);
+	if (ast_get_builtin_feature(chan, "atxfer", feature_code, sizeof(feature_code)) ||
+		ast_strlen_zero(feature_code)) {
+		pbx_builtin_setvar_helper(chan, "ATTENDEDTRANSFERSTATUS", "NOTPERMITTED");
+		ast_channel_unlock(chan);
+		return 0;
+	}
+	ast_channel_unlock(chan);
+
+	parse = ast_strdupa(data);
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	exten = args.exten;
+
+	for (digit = feature_code; *digit; ++digit) {
+		f.subclass.integer = *digit;
+		ast_queue_frame(chan, &f);
+	}
+
+	for (digit = exten; *digit; ++digit) {
+		f.subclass.integer = *digit;
+		ast_queue_frame(chan, &f);
+	}
+
+	f.subclass.integer = '#';
+	ast_queue_frame(chan, &f);
+
+	pbx_builtin_setvar_helper(chan, "ATTENDEDTRANSFERSTATUS", "SUCCESS");
+
+	return 0;
+}
+
+static int unload_module(void)
+{
+	return ast_unregister_application(app);
+}
+
+static int load_module(void)
+{
+	return ast_register_application_xml(app, attended_transfer_exec);
+}
+
+AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Attended transfer to the given extension");

apps/app_blind_transfer.c

@@ -0,0 +1,137 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2019, Alexei Gradinari
+ *
+ * Alexei Gradinari <alex2grad@gmail.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Blind transfer by caller channel
+ *
+ * \author Alexei Gradinari <alex2grad@gmail.com>
+ *
+ * \ingroup applications
+ */
+
+/*** MODULEINFO
+	<support_level>extended</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/app.h"
+#include "asterisk/channel.h"
+#include "asterisk/bridge.h"
+
+/*** DOCUMENTATION
+	<application name="BlindTransfer" language="en_US">
+		<synopsis>
+			Blind transfer channel(s) to the extension and context provided
+		</synopsis>
+		<syntax>
+			<parameter name="exten" required="true">
+				<para>Specify extension.</para>
+			</parameter>
+			<parameter name="context">
+				<para>Optionally specify a context.
+				By default, Asterisk will use the caller channel context.</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Redirect all channels currently bridged to the caller channel to the
+			specified destination.</para>
+			<para>The result of the application will be reported in the <variable>BLINDTRANSFERSTATUS</variable>
+			channel variable:</para>
+			<variablelist>
+				<variable name="BLINDTRANSFERSTATUS">
+					<value name="SUCCESS">
+						Transfer succeeded.
+					</value>
+					<value name="FAILURE">
+						Transfer failed.
+					</value>
+					<value name="INVALID">
+						Transfer invalid.
+					</value>
+					<value name="NOTPERMITTED">
+						Transfer not permitted.
+					</value>
+				</variable>
+			</variablelist>
+		</description>
+	</application>
+ ***/
+
+static const char * const app = "BlindTransfer";
+
+static int blind_transfer_exec(struct ast_channel *chan, const char *data)
+{
+	char *exten = NULL;
+	char *context = NULL;
+	char *parse;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(exten);
+		AST_APP_ARG(context);
+	);
+
+	if (ast_strlen_zero((char *)data)) {
+		ast_log(LOG_WARNING, "%s requires an argument (exten)\n", app);
+		pbx_builtin_setvar_helper(chan, "BLINDTRANSFERSTATUS", "FAILURE");
+		return 0;
+	}
+
+	parse = ast_strdupa(data);
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	exten = args.exten;
+	if (ast_strlen_zero(args.context)) {
+		context = (char *)ast_channel_context(chan);
+	} else {
+		context = args.context;
+	}
+
+	switch (ast_bridge_transfer_blind(1, chan, exten, context, NULL, NULL)) {
+		case AST_BRIDGE_TRANSFER_NOT_PERMITTED:
+			pbx_builtin_setvar_helper(chan, "BLINDTRANSFERSTATUS", "NOTPERMITTED");
+			break;
+		case AST_BRIDGE_TRANSFER_INVALID:
+			pbx_builtin_setvar_helper(chan, "BLINDTRANSFERSTATUS", "INVALID");
+			break;
+		case AST_BRIDGE_TRANSFER_FAIL:
+			pbx_builtin_setvar_helper(chan, "BLINDTRANSFERSTATUS", "FAILURE");
+			break;
+		case AST_BRIDGE_TRANSFER_SUCCESS:
+			pbx_builtin_setvar_helper(chan, "BLINDTRANSFERSTATUS", "SUCCESS");
+			break;
+		default:
+			pbx_builtin_setvar_helper(chan, "BLINDTRANSFERSTATUS", "FAILURE");
+        }
+
+	return 0;
+}
+
+static int unload_module(void)
+{
+	return ast_unregister_application(app);
+}
+
+static int load_module(void)
+{
+	return ast_register_application_xml(app, blind_transfer_exec);
+}
+
+AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Blind transfer channel to the given destination");

apps/app_bridgewait.c

@@ -100,6 +100,9 @@
 						<para>Automatically exit the bridge and return to the PBX after
 						<emphasis>duration</emphasis> seconds.</para>
 					</option>
+					<option name="n">
+						<para>Do not automatically answer the channel.</para>
+					</option>
 				</optionlist>
 			</parameter>
 		</syntax>
@@ -108,7 +111,7 @@
 			The channel will then wait in the holding bridge until some event occurs
 			which removes it from the holding bridge.</para>
 			<note><para>This application will answer calls which haven't already
-			been answered.</para></note>
+			been answered, unless the n option is provided.</para></note>
 		</description>
 	</application>
  ***/
@@ -186,6 +189,7 @@ enum bridgewait_flags {
 	MUXFLAG_MOHCLASS = (1 << 0),
 	MUXFLAG_ENTERTAINMENT = (1 << 1),
 	MUXFLAG_TIMEOUT = (1 << 2),
+	MUXFLAG_NOANSWER = (1 << 3),
 };
 
 enum bridgewait_args {
@@ -199,6 +203,7 @@ AST_APP_OPTIONS(bridgewait_opts, {
 	AST_APP_OPTION_ARG('e', MUXFLAG_ENTERTAINMENT, OPT_ARG_ENTERTAINMENT),
 	AST_APP_OPTION_ARG('m', MUXFLAG_MOHCLASS, OPT_ARG_MOHCLASS),
 	AST_APP_OPTION_ARG('S', MUXFLAG_TIMEOUT, OPT_ARG_TIMEOUT),
+	AST_APP_OPTION('n', MUXFLAG_NOANSWER),
 });
 
 static int bridgewait_timeout_callback(struct ast_bridge_channel *bridge_channel, void *hook_pvt)
@@ -391,17 +396,6 @@ static void wait_wrapper_removal(struct wait_bridge_wrapper *wrapper)
 	ao2_cleanup(wrapper);
 }
 
-/*!
- * \internal
- * \since 12.0.0
- * \brief Application callback for the bridgewait application
- *
- * \param chan channel running the application
- * \param data Arguments to the application
- *
- * \retval 0 Ran successfully and the call didn't hang up
- * \retval -1 Failed or the call was hung up by the time the channel exited the holding bridge
- */
 static enum wait_bridge_roles validate_role(const char *role)
 {
 	if (!strcmp(role, "participant")) {
@@ -413,6 +407,17 @@ static enum wait_bridge_roles validate_role(const char *role)
 	}
 }
 
+/*!
+ * \internal
+ * \since 12.0.0
+ * \brief Application callback for the bridgewait application
+ *
+ * \param chan channel running the application
+ * \param data Arguments to the application
+ *
+ * \retval 0 Ran successfully and the call didn't hang up
+ * \retval -1 Failed or the call was hung up by the time the channel exited the holding bridge
+ */
 static int bridgewait_exec(struct ast_channel *chan, const char *data)
 {
 	char *bridge_name = DEFAULT_BRIDGE_NAME;
@@ -458,7 +463,7 @@ static int bridgewait_exec(struct ast_channel *chan, const char *data)
 	}
 
 	/* Answer the channel if needed */
-	if (ast_channel_state(chan) != AST_STATE_UP) {
+	if (ast_channel_state(chan) != AST_STATE_UP && !ast_test_flag(&flags, MUXFLAG_NOANSWER)) {
 		ast_answer(chan);
 	}
 

apps/app_cdr.c

@@ -119,13 +119,13 @@ struct app_cdr_message_payload {
 	/*! The name of the channel to be manipulated */
 	const char *channel_name;
 	/*! Disable the CDR for this channel */
-	int disable:1;
+	unsigned int disable:1;
 	/*! Re-enable the CDR for this channel */
-	int reenable:1;
+	unsigned int reenable:1;
 	/*! Reset the CDR */
-	int reset:1;
+	unsigned int reset:1;
 	/*! If reseting the CDR, keep the variables */
-	int keep_variables:1;
+	unsigned int keep_variables:1;
 };
 
 static void appcdr_callback(void *data, struct stasis_subscription *sub, struct stasis_message *message)

apps/app_chanisavail.c

@@ -51,16 +51,18 @@ static const char app[] = "ChanIsAvail";
 			Check channel availability
 		</synopsis>
 		<syntax>
-			<parameter name="Technology/Resource" required="true" argsep="&amp;">
+			<parameter name="Technology/Resource" required="false" argsep="&amp;">
+				<argument name="Technology/Resource" required="true">
+					<para>Specification of the device(s) to check.  These must be in the format of
+					<literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
+					represents a particular channel driver, and <replaceable>Resource</replaceable>
+					represents a resource available to that particular channel driver.</para>
+				</argument>
 				<argument name="Technology2/Resource2" multiple="true">
 					<para>Optional extra devices to check</para>
 					<para>If you need more than one enter them as
 					Technology2/Resource2&amp;Technology3/Resource3&amp;.....</para>
 				</argument>
-				<para>Specification of the device(s) to check.  These must be in the format of
-				<literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
-				represents a particular channel driver, and <replaceable>Resource</replaceable>
-				represents a resource available to that particular channel driver.</para>
 			</parameter>
 			<parameter name="options" required="false">
 				<optionlist>
@@ -99,25 +101,28 @@ static const char app[] = "ChanIsAvail";
 
 static int chanavail_exec(struct ast_channel *chan, const char *data)
 {
-	int inuse=-1, option_state=0, string_compare=0, option_all_avail=0;
+	int inuse = -1;
+	int option_state = 0;
+	int string_compare = 0;
+	int option_all_avail = 0;
 	int status;
-	char *info, tmp[512], trychan[512], *peers, *tech, *number, *rest, *cur;
+	char *info;
+	char trychan[512];
+	char *rest;
+	char *tech;
+	char *number;
 	struct ast_str *tmp_availchan = ast_str_alloca(2048);
 	struct ast_str *tmp_availorig = ast_str_alloca(2048);
 	struct ast_str *tmp_availstat = ast_str_alloca(2048);
 	struct ast_str *tmp_availcause = ast_str_alloca(2048);
 	struct ast_channel *tempchan;
+	struct ast_custom_function *cdr_prop_func = ast_custom_function_find("CDR_PROP");
 	AST_DECLARE_APP_ARGS(args,
 		AST_APP_ARG(reqchans);
 		AST_APP_ARG(options);
 	);
 
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "ChanIsAvail requires an argument (DAHDI/1&DAHDI/2)\n");
-		return -1;
-	}
-
-	info = ast_strdupa(data);
+	info = ast_strdupa(data ?: "");
 
 	AST_STANDARD_APP_ARGS(args, info);
 
@@ -132,61 +137,68 @@ static int chanavail_exec(struct ast_channel *chan, const char *data)
 			string_compare = 1;
 		}
 	}
-	peers = args.reqchans;
-	if (peers) {
-		cur = peers;
-		do {
-			/* remember where to start next time */
-			rest = strchr(cur, '&');
-			if (rest) {
-				*rest = 0;
-				rest++;
+
+	rest = args.reqchans;
+	if (!rest) {
+		rest = "";
+	}
+	while ((tech = strsep(&rest, "&"))) {
+		tech = ast_strip(tech);
+
+		number = strchr(tech, '/');
+		if (!number) {
+			if (!ast_strlen_zero(tech)) {
+				ast_log(LOG_WARNING, "Invalid ChanIsAvail technology/resource argument: '%s'\n",
+					tech);
 			}
-			tech = cur;
-			number = strchr(tech, '/');
-			if (!number) {
-				ast_log(LOG_WARNING, "ChanIsAvail argument takes format ([technology]/[device])\n");
-				return -1;
+
+			ast_str_append(&tmp_availstat, 0, "%s%d",
+				ast_str_strlen(tmp_availstat) ? "&" : "", AST_DEVICE_INVALID);
+			continue;
+		}
+		*number++ = '\0';
+
+		status = AST_DEVICE_UNKNOWN;
+
+		if (string_compare) {
+			/* ast_parse_device_state checks for "SIP/1234" as a channel name.
+			   ast_device_state will ask the SIP driver for the channel state. */
+
+			snprintf(trychan, sizeof(trychan), "%s/%s", tech, number);
+			status = inuse = ast_parse_device_state(trychan);
+		} else if (option_state) {
+			/* If the pbx says in use then don't bother trying further.
+			   This is to permit testing if someone's on a call, even if the
+			   channel can permit more calls (ie callwaiting, sip calls, etc).  */
+
+			snprintf(trychan, sizeof(trychan), "%s/%s", tech, number);
+			status = inuse = ast_device_state(trychan);
+		}
+		ast_str_append(&tmp_availstat, 0, "%s%d",
+			ast_str_strlen(tmp_availstat) ? "&" : "", status);
+		if ((inuse <= (int) AST_DEVICE_NOT_INUSE)
+			&& (tempchan = ast_request(tech, ast_channel_nativeformats(chan), NULL, chan, number, &status))) {
+			ast_str_append(&tmp_availchan, 0, "%s%s",
+				ast_str_strlen(tmp_availchan) ? "&" : "", ast_channel_name(tempchan));
+
+			ast_str_append(&tmp_availorig, 0, "%s%s/%s",
+				ast_str_strlen(tmp_availorig) ? "&" : "", tech, number);
+
+			ast_str_append(&tmp_availcause, 0, "%s%d",
+				ast_str_strlen(tmp_availcause) ? "&" : "", status);
+
+			/* Disable CDR for this temporary channel. */
+			if (cdr_prop_func) {
+				ast_func_write(tempchan, "CDR_PROP(disable)", "1");
 			}
-			*number = '\0';
-			number++;
 
-			status = AST_DEVICE_UNKNOWN;
+			ast_hangup(tempchan);
+			tempchan = NULL;
 
-			if (string_compare) {
-				/* ast_parse_device_state checks for "SIP/1234" as a channel name.
-				   ast_device_state will ask the SIP driver for the channel state. */
-
-				snprintf(trychan, sizeof(trychan), "%s/%s",cur,number);
-				status = inuse = ast_parse_device_state(trychan);
-			} else if (option_state) {
-				/* If the pbx says in use then don't bother trying further.
-				   This is to permit testing if someone's on a call, even if the
-				   channel can permit more calls (ie callwaiting, sip calls, etc).  */
-
-				snprintf(trychan, sizeof(trychan), "%s/%s",cur,number);
-				status = inuse = ast_device_state(trychan);
+			if (!option_all_avail) {
+				break;
 			}
-			snprintf(tmp, sizeof(tmp), "%d", status);
-			ast_str_append(&tmp_availstat, 0, "%s%s", ast_str_strlen(tmp_availstat) ? "&" : "", tmp);
-			if ((inuse <= 1) && (tempchan = ast_request(tech, ast_channel_nativeformats(chan), NULL, chan, number, &status))) {
-					ast_str_append(&tmp_availchan, 0, "%s%s", ast_str_strlen(tmp_availchan) ? "&" : "", ast_channel_name(tempchan));
-
-					snprintf(tmp, sizeof(tmp), "%s/%s", tech, number);
-					ast_str_append(&tmp_availorig, 0, "%s%s", ast_str_strlen(tmp_availorig) ? "&" : "", tmp);
-
-					snprintf(tmp, sizeof(tmp), "%d", status);
-					ast_str_append(&tmp_availcause, 0, "%s%s", ast_str_strlen(tmp_availcause) ? "&" : "", tmp);
-
-					ast_hangup(tempchan);
-					tempchan = NULL;
-
-					if (!option_all_avail) {
-						break;
-					}
-			}
-			cur = rest;
-		} while (cur);
+		}
 	}
 
 	pbx_builtin_setvar_helper(chan, "AVAILCHAN", ast_str_buffer(tmp_availchan));
@@ -209,4 +221,9 @@ static int load_module(void)
 		AST_MODULE_LOAD_DECLINE : AST_MODULE_LOAD_SUCCESS;
 }
 
-AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Check channel availability");
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Check channel availability",
+	.support_level = AST_MODULE_SUPPORT_EXTENDED,
+	.load = load_module,
+	.unload = unload_module,
+	.optional_modules = "func_cdr"
+);

apps/app_chanspy.c

@@ -509,7 +509,8 @@ static int start_spying(struct ast_autochan *autochan, const char *spychan_name,
 	int res;
 
 	ast_autochan_channel_lock(autochan);
-	ast_log(LOG_NOTICE, "Attaching %s to %s\n", spychan_name, ast_channel_name(autochan->chan));
+	ast_verb(3, "Attaching spy channel %s to %s\n",
+		spychan_name, ast_channel_name(autochan->chan));
 
 	if (ast_test_flag(flags, OPTION_READONLY)) {
 		ast_set_flag(audiohook, AST_AUDIOHOOK_MUTE_WRITE);
@@ -558,7 +559,7 @@ static int pack_channel_into_message(struct ast_channel *chan, const char *role,
  * \brief Publish the chanspy message over Stasis-Core
  * \param spyer The channel doing the spying
  * \param spyee Who is being spied upon
- * \start start If non-zero, the spying is starting. Otherwise, the spyer is
+ * \param start If non-zero, the spying is starting. Otherwise, the spyer is
  * finishing
  */
 static void publish_chanspy_message(struct ast_channel *spyer,
@@ -844,13 +845,13 @@ static int channel_spy(struct ast_channel *chan, struct ast_autochan *spyee_auto
 	ast_audiohook_unlock(&csth.spy_audiohook);
 	ast_audiohook_destroy(&csth.spy_audiohook);
 
+	ast_verb(2, "Done Spying on channel %s\n", name);
+	publish_chanspy_message(chan, spyee_autochan->chan, 0);
+
 	if (spyee_bridge_autochan) {
 		ast_autochan_destroy(spyee_bridge_autochan);
 	}
 
-	ast_verb(2, "Done Spying on channel %s\n", name);
-	publish_chanspy_message(chan, NULL, 0);
-
 	return running;
 }
 
@@ -898,7 +899,7 @@ static int common_exec(struct ast_channel *chan, struct ast_flags *flags,
 	signed char zero_volume = 0;
 	int waitms;
 	int res;
-	int num_spyed_upon = 1;
+	int num_spied_upon = 1;
 	struct ast_channel_iterator *iter = NULL;
 
 	if (ast_test_flag(flags, OPTION_EXIT)) {
@@ -925,7 +926,7 @@ static int common_exec(struct ast_channel *chan, struct ast_flags *flags,
 		struct ast_autochan *autochan = NULL, *next_autochan = NULL;
 		struct ast_channel *prev = NULL;
 
-		if (!ast_test_flag(flags, OPTION_QUIET) && num_spyed_upon) {
+		if (!ast_test_flag(flags, OPTION_QUIET) && num_spied_upon) {
 			res = ast_streamfile(chan, "beep", ast_channel_language(chan));
 			if (!res)
 				res = ast_waitstream(chan, "");
@@ -990,7 +991,7 @@ static int common_exec(struct ast_channel *chan, struct ast_flags *flags,
 
 		/* reset for the next loop around, unless overridden later */
 		waitms = 100;
-		num_spyed_upon = 0;
+		num_spied_upon = 0;
 
 		for (autochan = next_channel(iter, chan);
 			autochan;
@@ -1145,7 +1146,7 @@ static int common_exec(struct ast_channel *chan, struct ast_flags *flags,
 			}
 
 			res = channel_spy(chan, autochan, &volfactor, fd, user_options, flags, exitcontext);
-			num_spyed_upon++;
+			num_spied_upon++;
 
 			if (res == -1) {
 				ast_autochan_destroy(autochan);

apps/app_confbridge.c

@@ -124,10 +124,48 @@
 				</variable>
 			</variablelist>
 		</description>
+		<see-also>
+			<ref type="application">ConfKick</ref>
+			<ref type="function">CONFBRIDGE</ref>
+			<ref type="function">CONFBRIDGE_INFO</ref>
+			<ref type="function">CONFBRIDGE_CHANNELS</ref>
+		</see-also>
+	</application>
+	<application name="ConfKick" language="en_US">
+		<since>
+			<version>16.19.0</version>
+			<version>18.5.0</version>
+			<version>19.0.0</version>
+		</since>
+		<synopsis>
+			Kicks channel(s) from the requested ConfBridge.
+		</synopsis>
+		<syntax>
+			<parameter name="conference" required="true" />
+			<parameter name="channel">
+				<para>The channel to kick, <literal>all</literal>
+				to kick all users, or <literal>participants</literal>
+				to kick all non-admin participants. Default is all.</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Kicks the requested channel(s) from a conference bridge.</para>
+			<variablelist>
+				<variable name="CONFKICKSTATUS">
+					<value name="FAILURE">
+						Could not kick any users with the provided arguments.
+					</value>
+					<value name="SUCCESS">
+						Successfully kicked users from specified conference bridge.
+					</value>
+				</variable>
+			</variablelist>
+		</description>
 		<see-also>
 			<ref type="application">ConfBridge</ref>
 			<ref type="function">CONFBRIDGE</ref>
 			<ref type="function">CONFBRIDGE_INFO</ref>
+			<ref type="function">CONFBRIDGE_CHANNELS</ref>
 		</see-also>
 	</application>
 	<function name="CONFBRIDGE" language="en_US">
@@ -160,23 +198,29 @@
 			<para>---- Example 1 ----</para>
 			<para>In this example the custom user profile set on the channel will
 			automatically be used by the ConfBridge application.</para>
-			<para>exten => 1,1,Answer()</para>
+			<example title="Example 1">
+			exten => 1,1,Answer()
+			</example>
 			<para>; In this example the effect of the following line is</para>
 			<para>; implied:</para>
-			<para>; same => n,Set(CONFBRIDGE(user,template)=default_user)</para>
-			<para>same => n,Set(CONFBRIDGE(user,announce_join_leave)=yes)</para>
-			<para>same => n,Set(CONFBRIDGE(user,startmuted)=yes)</para>
-			<para>same => n,ConfBridge(1) </para>
+			<example title="Example 1b">
+			same => n,Set(CONFBRIDGE(user,template)=default_user)
+			same => n,Set(CONFBRIDGE(user,announce_join_leave)=yes)
+			same => n,Set(CONFBRIDGE(user,startmuted)=yes)
+			same => n,ConfBridge(1)
+			</example>
 			<para>---- Example 2 ----</para>
 			<para>This example shows how to use a predefined user profile in
 			<filename>confbridge.conf</filename> as a template for a dynamic profile.
 			Here we make an admin/marked user out of the <literal>my_user</literal>
 			profile that you define in <filename>confbridge.conf</filename>.</para>
-			<para>exten => 1,1,Answer()</para>
-			<para>same => n,Set(CONFBRIDGE(user,template)=my_user)</para>
-			<para>same => n,Set(CONFBRIDGE(user,admin)=yes)</para>
-			<para>same => n,Set(CONFBRIDGE(user,marked)=yes)</para>
-			<para>same => n,ConfBridge(1)</para>
+			<example title="Example 2">
+			exten => 1,1,Answer()
+			same => n,Set(CONFBRIDGE(user,template)=my_user)
+			same => n,Set(CONFBRIDGE(user,admin)=yes)
+			same => n,Set(CONFBRIDGE(user,marked)=yes)
+			same => n,ConfBridge(1)
+			</example>
 		</description>
 	</function>
 	<function name="CONFBRIDGE_INFO" language="en_US">
@@ -212,6 +256,50 @@
 			<para>This function returns a non-negative integer for valid conference
 			names and an empty string for invalid conference names.</para>
 		</description>
+		<see-also>
+			<ref type="function">CONFBRIDGE_CHANNELS</ref>
+		</see-also>
+	</function>
+	<function name="CONFBRIDGE_CHANNELS" language="en_US">
+		<since>
+			<version>16.26.0</version>
+			<version>18.12.0</version>
+			<version>19.4.0</version>
+		</since>
+		<synopsis>
+			Get a list of channels in a ConfBridge conference.
+		</synopsis>
+		<syntax>
+			<parameter name="type" required="true">
+				<para>What conference information is requested.</para>
+				<enumlist>
+					<enum name="admins">
+						<para>Get the number of admin users in the conference.</para>
+					</enum>
+					<enum name="marked">
+						<para>Get the number of marked users in the conference.</para>
+					</enum>
+					<enum name="parties">
+						<para>Get the number of total users in the conference.</para>
+					</enum>
+					<enum name="active">
+						<para>Get the number of active users in the conference.</para>
+					</enum>
+					<enum name="waiting">
+						<para>Get the number of waiting users in the conference.</para>
+					</enum>
+				</enumlist>
+			</parameter>
+			<parameter name="conf" required="true">
+				<para>The name of the conference being referenced.</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>This function returns a comma-separated list of channels in a ConfBridge conference, optionally filtered by a type of participant.</para>
+		</description>
+		<see-also>
+			<ref type="function">CONFBRIDGE_INFO</ref>
+		</see-also>
 	</function>
 	<manager name="ConfbridgeList" language="en_US">
 		<synopsis>
@@ -305,6 +393,37 @@
 				ConfbridgeListRoomsComplete.</para>
 		</description>
 	</manager>
+	<managerEvent language="en_US" name="ConfbridgeListRooms">
+		<managerEventInstance class="EVENT_FLAG_REPORTING">
+			<synopsis>Raised as part of the ConfbridgeListRooms action response list.</synopsis>
+			<syntax>
+				<parameter name="Conference">
+					<para>The name of the Confbridge conference.</para>
+				</parameter>
+				<parameter name="Parties">
+					<para>Number of users in the conference.</para>
+					<para>This includes both active and waiting users.</para>
+				</parameter>
+				<parameter name="Marked">
+					<para>Number of marked users in the conference.</para>
+				</parameter>
+				<parameter name="Locked">
+					<para>Is the conference locked?</para>
+					<enumlist>
+						<enum name="Yes"/>
+						<enum name="No"/>
+					</enumlist>
+				</parameter>
+				<parameter name="Muted">
+					<para>Is the conference muted?</para>
+					<enumlist>
+						<enum name="Yes"/>
+						<enum name="No"/>
+					</enumlist>
+				</parameter>
+			</syntax>
+		</managerEventInstance>
+	</managerEvent>
 	<manager name="ConfbridgeMute" language="en_US">
 		<synopsis>
 			Mute a Confbridge user.
@@ -426,6 +545,7 @@
  */
 
 static const char app[] = "ConfBridge";
+static const char app2[] = "ConfKick";
 
 /*! Number of buckets our conference bridges container can have */
 #define CONFERENCE_BRIDGE_BUCKETS 53
@@ -576,6 +696,10 @@ static void send_conf_stasis(struct confbridge_conference *conference, struct as
 		return;
 	}
 
+	if (ast_test_flag(&conference->b_profile, BRIDGE_OPT_ENABLE_EVENTS)) {
+		conf_send_event_to_participants(conference, chan, msg);
+	}
+
 	if (channel_topic) {
 		stasis_publish(ast_channel_topic(chan), msg);
 	} else {
@@ -583,6 +707,43 @@ static void send_conf_stasis(struct confbridge_conference *conference, struct as
 	}
 }
 
+static void send_conf_stasis_snapshots(struct confbridge_conference *conference,
+	struct ast_channel_snapshot *chan_snapshot, struct stasis_message_type *type,
+	struct ast_json *extras)
+{
+	RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup);
+	RAII_VAR(struct ast_json *, json_object, NULL, ast_json_unref);
+	RAII_VAR(struct ast_bridge_snapshot *, bridge_snapshot, NULL, ao2_cleanup);
+
+	json_object = ast_json_pack("{s: s}",
+		"conference", conference->name);
+	if (!json_object) {
+		return;
+	}
+
+	if (extras) {
+		ast_json_object_update(json_object, extras);
+	}
+
+	ast_bridge_lock(conference->bridge);
+	bridge_snapshot = ast_bridge_snapshot_create(conference->bridge);
+	ast_bridge_unlock(conference->bridge);
+	if (!bridge_snapshot) {
+		return;
+	}
+
+	msg = ast_bridge_blob_create_from_snapshots(type,
+		bridge_snapshot,
+		chan_snapshot,
+		json_object);
+	if (!msg) {
+		return;
+	}
+
+	stasis_publish(ast_bridge_topic(conference->bridge), msg);
+}
+
+
 static void send_conf_start_event(struct confbridge_conference *conference)
 {
 	send_conf_stasis(conference, NULL, confbridge_start_type(), NULL, 0);
@@ -724,8 +885,8 @@ struct confbridge_conference *conf_find_bridge(const char *conference_name)
  *
  * \note Must be called with the conference locked
  *
- * \retval 1, conference is recording.
- * \retval 0, conference is NOT recording.
+ * \retval 1 conference is recording.
+ * \retval 0 conference is NOT recording.
  */
 static int conf_is_recording(struct confbridge_conference *conference)
 {
@@ -836,7 +997,7 @@ static int conf_start_record(struct confbridge_conference *conference)
 	return 0;
 }
 
-/* \brief Playback the given filename and monitor for any dtmf interrupts.
+/*! \brief Playback the given filename and monitor for any dtmf interrupts.
  *
  * This function is used to playback sound files on a given channel and optionally
  * allow dtmf interrupts to occur.
@@ -850,7 +1011,9 @@ static int conf_start_record(struct confbridge_conference *conference)
  * \param channel Optional channel to play file on if bridge_channel not given
  * \param filename The file name to playback
  *
- * \retval -1 failure during playback, 0 on file was fully played, 1 on dtmf interrupt.
+ * \retval -1 failure during playback.
+ * \retval 0 on file was fully played.
+ * \retval 1 on dtmf interrupt.
  */
 static int play_file(struct ast_bridge_channel *bridge_channel, struct ast_channel *channel,
 		     const char *filename)
@@ -907,7 +1070,8 @@ static int sound_file_exists(const char *filename)
  * \param bridge_channel The bridged channel involved
  *
  * \note if caller is NULL, the announcment will be sent to all participants in the conference.
- * \return Returns 0 on success, -1 if the user hung up
+ * \retval 0 on success.
+ * \retval -1 if the user hung up.
  */
 static int announce_user_count(struct confbridge_conference *conference, struct confbridge_user *user,
 			       struct ast_bridge_channel *bridge_channel)
@@ -957,7 +1121,9 @@ static int announce_user_count(struct confbridge_conference *conference, struct
  * \param user User to play audio prompt to
  * \param filename Prompt to play
  *
- * \return Returns 0 on success, -1 if the user hung up
+ * \retval 0 on success.
+ * \retval -1 if the user hung up.
+ *
  * \note Generally this should be called when the conference is unlocked to avoid blocking
  * the entire conference while the sound is played. But don't unlock the conference bridge
  * in the middle of a state transition.
@@ -1094,8 +1260,6 @@ static void hangup_data_destroy(struct hangup_data *hangup)
  * \brief Destroy a conference bridge
  *
  * \param obj The conference bridge object
- *
- * \return Returns nothing
  */
 static void destroy_conference_bridge(void *obj)
 {
@@ -1107,13 +1271,15 @@ static void destroy_conference_bridge(void *obj)
 		if (conference->playback_queue) {
 			struct hangup_data hangup;
 			hangup_data_init(&hangup, conference);
-			ast_taskprocessor_push(conference->playback_queue, hangup_playback, &hangup);
 
-			ast_mutex_lock(&hangup.lock);
-			while (!hangup.hungup) {
-				ast_cond_wait(&hangup.cond, &hangup.lock);
+			if (!ast_taskprocessor_push(conference->playback_queue, hangup_playback, &hangup)) {
+				ast_mutex_lock(&hangup.lock);
+				while (!hangup.hungup) {
+					ast_cond_wait(&hangup.cond, &hangup.lock);
+				}
+				ast_mutex_unlock(&hangup.lock);
 			}
-			ast_mutex_unlock(&hangup.lock);
+
 			hangup_data_destroy(&hangup);
 		} else {
 			/* Playback queue is not yet allocated. Just hang up the channel straight */
@@ -1230,7 +1396,7 @@ void conf_update_user_mute(struct confbridge_user *user)
 		ast_channel_name(user->chan));
 }
 
-/*
+/*!
  * \internal
  * \brief Mute/unmute a single user.
  */
@@ -1314,8 +1480,6 @@ void conf_moh_start(struct confbridge_user *user)
  * \brief Unsuspend MOH for the conference user.
  *
  * \param user Conference user to unsuspend MOH on.
- *
- * \return Nothing
  */
 static void conf_moh_unsuspend(struct confbridge_user *user)
 {
@@ -1331,8 +1495,6 @@ static void conf_moh_unsuspend(struct confbridge_user *user)
  * \brief Suspend MOH for the conference user.
  *
  * \param user Conference user to suspend MOH on.
- *
- * \return Nothing
  */
 static void conf_moh_suspend(struct confbridge_user *user)
 {
@@ -1354,7 +1516,7 @@ int conf_handle_inactive_waitmarked(struct confbridge_user *user)
 	return 0;
 }
 
-int conf_handle_only_unmarked(struct confbridge_user *user)
+int conf_handle_only_person(struct confbridge_user *user)
 {
 	/* If audio prompts have not been quieted or this prompt quieted play it on out */
 	if (!ast_test_flag(&user->u_profile, USER_OPT_QUIET | USER_OPT_NOONLYPERSON)) {
@@ -1473,6 +1635,126 @@ static int push_announcer(struct confbridge_conference *conference)
 	return 0;
 }
 
+static void confbridge_unlock_and_unref(void *obj)
+{
+	struct confbridge_conference *conference = obj;
+
+	if (!obj) {
+		return;
+	}
+	ao2_unlock(conference);
+	ao2_ref(conference, -1);
+}
+
+void confbridge_handle_atxfer(struct ast_attended_transfer_message *msg)
+{
+	struct ast_channel_snapshot *old_snapshot;
+	struct ast_channel_snapshot *new_snapshot;
+	char *confbr_name = NULL;
+	char *comma;
+	RAII_VAR(struct confbridge_conference *, conference, NULL, confbridge_unlock_and_unref);
+	struct confbridge_user *user = NULL;
+	int found_user = 0;
+	struct ast_json *json_object;
+
+	if (msg->to_transferee.channel_snapshot
+		&& strcmp(msg->to_transferee.channel_snapshot->appl, "ConfBridge") == 0
+		&& msg->target) {
+		/* We're transferring a bridge to an extension */
+		old_snapshot = msg->to_transferee.channel_snapshot;
+		new_snapshot = msg->target;
+	} else if (msg->to_transfer_target.channel_snapshot
+		&& strcmp(msg->to_transfer_target.channel_snapshot->appl, "ConfBridge") == 0
+		&& msg->transferee) {
+		/* We're transferring a call to a bridge */
+		old_snapshot = msg->to_transfer_target.channel_snapshot;
+		new_snapshot = msg->transferee;
+	} else {
+		ast_log(LOG_ERROR, "Could not determine proper channels\n");
+		return;
+	}
+
+	/*
+	 * old_snapshot->data should have the original parameters passed to
+	 * the ConfBridge app:
+	 * conference[,bridge_profile[,user_profile[,menu]]]
+	 * We'll use "conference" to look up the bridge.
+	 *
+	 * We _could_ use old_snapshot->bridgeid to get the bridge but
+	 * that would involve locking the conference_bridges container
+	 * and iterating over it looking for a matching bridge.
+	 */
+	if (ast_strlen_zero(old_snapshot->data)) {
+		ast_log(LOG_ERROR, "Channel '%s' didn't have app data set\n", old_snapshot->name);
+		return;
+	}
+	confbr_name = ast_strdupa(old_snapshot->data);
+	comma = strchr(confbr_name, ',');
+	if (comma) {
+		*comma = '\0';
+	}
+
+	ast_debug(1, "Confbr: %s  Leaving: %s  Joining: %s\n", confbr_name, old_snapshot->name, new_snapshot->name);
+
+	conference = ao2_find(conference_bridges, confbr_name, OBJ_SEARCH_KEY);
+	if (!conference) {
+		ast_log(LOG_ERROR, "Conference bridge '%s' not found\n", confbr_name);
+		return;
+	}
+	ao2_lock(conference);
+
+	/*
+	 * We need to grab the user profile for the departing user in order to
+	 * properly format the join/leave messages.
+	 */
+	AST_LIST_TRAVERSE(&conference->active_list, user, list) {
+		if (strcasecmp(ast_channel_name(user->chan), old_snapshot->name) == 0) {
+			found_user = 1;
+			break;
+		}
+	}
+
+	/*
+	 * If we didn't find the user in the active list, try the waiting list.
+	 */
+	if (!found_user && conference->waitingusers) {
+		AST_LIST_TRAVERSE(&conference->waiting_list, user, list) {
+			if (strcasecmp(ast_channel_name(user->chan), old_snapshot->name) == 0) {
+				found_user = 1;
+				break;
+			}
+		}
+	}
+
+	if (!found_user) {
+		ast_log(LOG_ERROR, "Unable to find user profile for channel '%s' in bridge '%s'\n",
+			old_snapshot->name, confbr_name);
+		return;
+	}
+
+	/*
+	 * We're going to use the existing user profile to create the messages.
+	 */
+	json_object = ast_json_pack("{s: b}",
+		"admin", ast_test_flag(&user->u_profile, USER_OPT_ADMIN)
+	);
+	if (!json_object) {
+		return;
+	}
+
+	send_conf_stasis_snapshots(conference, old_snapshot, confbridge_leave_type(), json_object);
+	ast_json_unref(json_object);
+
+	json_object = ast_json_pack("{s: b, s: b}",
+		"admin", ast_test_flag(&user->u_profile, USER_OPT_ADMIN),
+		"muted", user->muted);
+	if (!json_object) {
+		return;
+	}
+	send_conf_stasis_snapshots(conference, new_snapshot, confbridge_join_type(), json_object);
+	ast_json_unref(json_object);
+}
+
 /*!
  * \brief Join a conference bridge
  *
@@ -1487,7 +1769,7 @@ static struct confbridge_conference *join_conference_bridge(const char *conferen
 	struct post_join_action *action;
 	int max_members_reached = 0;
 
-	/* We explictly lock the conference bridges container ourselves so that other callers can not create duplicate conferences at the same */
+	/* We explicitly lock the conference bridges container ourselves so that other callers can not create duplicate conferences at the same time */
 	ao2_lock(conference_bridges);
 
 	ast_debug(1, "Trying to find conference bridge '%s'\n", conference_name);
@@ -1543,6 +1825,8 @@ static struct confbridge_conference *join_conference_bridge(const char *conferen
 
 		/* Set the internal sample rate on the bridge from the bridge profile */
 		ast_bridge_set_internal_sample_rate(conference->bridge, conference->b_profile.internal_sample_rate);
+		/* Set the maximum sample rate on the bridge from the bridge profile */
+		ast_bridge_set_maximum_sample_rate(conference->bridge, conference->b_profile.maximum_sample_rate);
 		/* Set the internal mixing interval on the bridge from the bridge profile */
 		ast_bridge_set_mixing_interval(conference->bridge, conference->b_profile.mix_interval);
 		ast_bridge_set_binaural_active(conference->bridge, ast_test_flag(&conference->b_profile, BRIDGE_OPT_BINAURAL_ACTIVE));
@@ -1551,7 +1835,6 @@ static struct confbridge_conference *join_conference_bridge(const char *conferen
 			ast_bridge_set_talker_src_video_mode(conference->bridge);
 		} else if (ast_test_flag(&conference->b_profile, BRIDGE_OPT_VIDEO_SRC_SFU)) {
 			ast_bridge_set_sfu_video_mode(conference->bridge);
-			ast_bridge_set_video_update_discard(conference->bridge, conference->b_profile.video_update_discard);
 			ast_bridge_set_remb_send_interval(conference->bridge, conference->b_profile.remb_send_interval);
 			if (ast_test_flag(&conference->b_profile, BRIDGE_OPT_REMB_BEHAVIOR_AVERAGE)) {
 				ast_brige_set_remb_behavior(conference->bridge, AST_BRIDGE_VIDEO_SFU_REMB_AVERAGE);
@@ -1559,9 +1842,25 @@ static struct confbridge_conference *join_conference_bridge(const char *conferen
 				ast_brige_set_remb_behavior(conference->bridge, AST_BRIDGE_VIDEO_SFU_REMB_LOWEST);
 			} else if (ast_test_flag(&conference->b_profile, BRIDGE_OPT_REMB_BEHAVIOR_HIGHEST)) {
 				ast_brige_set_remb_behavior(conference->bridge, AST_BRIDGE_VIDEO_SFU_REMB_HIGHEST);
+			} else if (ast_test_flag(&conference->b_profile, BRIDGE_OPT_REMB_BEHAVIOR_AVERAGE_ALL)) {
+				ast_brige_set_remb_behavior(conference->bridge, AST_BRIDGE_VIDEO_SFU_REMB_AVERAGE_ALL);
+			} else if (ast_test_flag(&conference->b_profile, BRIDGE_OPT_REMB_BEHAVIOR_LOWEST_ALL)) {
+				ast_brige_set_remb_behavior(conference->bridge, AST_BRIDGE_VIDEO_SFU_REMB_LOWEST_ALL);
+			} else if (ast_test_flag(&conference->b_profile, BRIDGE_OPT_REMB_BEHAVIOR_HIGHEST_ALL)) {
+				ast_brige_set_remb_behavior(conference->bridge, AST_BRIDGE_VIDEO_SFU_REMB_HIGHEST_ALL);
+			} else if (ast_test_flag(&conference->b_profile, BRIDGE_OPT_REMB_BEHAVIOR_FORCE)) {
+				ast_brige_set_remb_behavior(conference->bridge, AST_BRIDGE_VIDEO_SFU_REMB_FORCE);
+				ast_bridge_set_remb_estimated_bitrate(conference->bridge, conference->b_profile.remb_estimated_bitrate);
 			}
 		}
 
+		/* Always set the minimum interval between video updates, to avoid infinite video updates. */
+		ast_bridge_set_video_update_discard(conference->bridge, conference->b_profile.video_update_discard);
+
+		if (ast_test_flag(&conference->b_profile, BRIDGE_OPT_ENABLE_EVENTS)) {
+			ast_bridge_set_send_sdp_label(conference->bridge, 1);
+		}
+
 		/* Link it into the conference bridges container */
 		if (!ao2_link(conference_bridges, conference)) {
 			ao2_ref(conference, -1);
@@ -2311,6 +2610,25 @@ static int join_callback(struct ast_bridge_channel *bridge_channel, void *ignore
 	return 0;
 }
 
+struct confbridge_hook_data {
+	struct confbridge_conference *conference;
+	struct confbridge_user *user;
+	enum ast_bridge_hook_type hook_type;
+};
+
+static int send_event_hook_callback(struct ast_bridge_channel *bridge_channel, void *data)
+{
+	struct confbridge_hook_data *hook_data = data;
+
+	if (hook_data->hook_type == AST_BRIDGE_HOOK_TYPE_JOIN) {
+		send_join_event(hook_data->user, hook_data->conference);
+	} else {
+		send_leave_event(hook_data->user, hook_data->conference);
+	}
+
+	return 0;
+}
+
 /*! \brief The ConfBridge application */
 static int confbridge_exec(struct ast_channel *chan, const char *data)
 {
@@ -2328,6 +2646,9 @@ static int confbridge_exec(struct ast_channel *chan, const char *data)
 		.tech_args.silence_threshold = DEFAULT_SILENCE_THRESHOLD,
 		.tech_args.drop_silence = 0,
 	};
+	struct confbridge_hook_data *join_hook_data;
+	struct confbridge_hook_data *leave_hook_data;
+
 	AST_DECLARE_APP_ARGS(args,
 		AST_APP_ARG(conf_name);
 		AST_APP_ARG(b_profile_name);
@@ -2335,10 +2656,6 @@ static int confbridge_exec(struct ast_channel *chan, const char *data)
 		AST_APP_ARG(menu_profile_name);
 	);
 
-	if (ast_channel_state(chan) != AST_STATE_UP) {
-		ast_answer(chan);
-	}
-
 	if (ast_bridge_features_init(&user.features)) {
 		pbx_builtin_setvar_helper(chan, "CONFBRIDGE_RESULT", "FAILED");
 		res = -1;
@@ -2388,6 +2705,11 @@ static int confbridge_exec(struct ast_channel *chan, const char *data)
 		goto confbridge_cleanup;
 	}
 
+	/* If channel hasn't been answered already, answer it, unless we're explicitly not supposed to */
+	if ((ast_channel_state(chan) != AST_STATE_UP) && (ast_test_flag(&user.u_profile, USER_OPT_ANSWER_CHANNEL))) {
+		ast_answer(chan);
+	}
+
 	quiet = ast_test_flag(&user.u_profile, USER_OPT_QUIET);
 
 	/* ask for a PIN immediately after finding user profile.  This has to be
@@ -2431,6 +2753,13 @@ static int confbridge_exec(struct ast_channel *chan, const char *data)
 		user.features.dtmf_passthrough = 0;
 	}
 
+	/* Set if text messaging is enabled for this user or not */
+	if (ast_test_flag(&user.u_profile, USER_OPT_TEXT_MESSAGING)) {
+		user.features.text_messaging = 1;
+	} else {
+		user.features.text_messaging = 0;
+	}
+
 	/* Set dsp threshold values if present */
 	if (user.u_profile.talking_threshold) {
 		user.tech_args.talking_threshold = user.u_profile.talking_threshold;
@@ -2481,17 +2810,24 @@ static int confbridge_exec(struct ast_channel *chan, const char *data)
 		ast_autoservice_stop(chan);
 	}
 
-	/* Play the Join sound to both the conference and the user entering. */
 	if (!quiet) {
 		const char *join_sound = conf_get_sound(CONF_SOUND_JOIN, conference->b_profile.sounds);
 
-		if (strcmp(conference->b_profile.language, ast_channel_language(chan))) {
-			ast_stream_and_wait(chan, join_sound, "");
+		/* if hear_own_join_sound is enabled play the Join sound to everyone */
+		if (ast_test_flag(&user.u_profile, USER_OPT_HEAR_OWN_JOIN_SOUND) ) {
+			if (strcmp(conference->b_profile.language, ast_channel_language(chan))) {
+				ast_stream_and_wait(chan, join_sound, "");
+				ast_autoservice_start(chan);
+				play_sound_file(conference, join_sound);
+				ast_autoservice_stop(chan);
+			} else {
+				async_play_sound_file(conference, join_sound, chan);
+			}
+		/* if hear_own_join_sound is disabled only play the Join sound to just the conference */
+		} else {
 			ast_autoservice_start(chan);
 			play_sound_file(conference, join_sound);
 			ast_autoservice_stop(chan);
-		} else {
-			async_play_sound_file(conference, join_sound, chan);
 		}
 	}
 
@@ -2510,8 +2846,39 @@ static int confbridge_exec(struct ast_channel *chan, const char *data)
 
 	conf_moh_unsuspend(&user);
 
-	/* Join our conference bridge for real */
-	send_join_event(&user, conference);
+	join_hook_data = ast_malloc(sizeof(*join_hook_data));
+	if (!join_hook_data) {
+		res = -1;
+		goto confbridge_cleanup;
+	}
+	join_hook_data->user = &user;
+	join_hook_data->conference = conference;
+	join_hook_data->hook_type = AST_BRIDGE_HOOK_TYPE_JOIN;
+	res = ast_bridge_join_hook(&user.features, send_event_hook_callback,
+		join_hook_data, ast_free_ptr, 0);
+	if (res) {
+		ast_free(join_hook_data);
+		ast_log(LOG_ERROR, "Couldn't add bridge join hook for channel '%s'\n", ast_channel_name(chan));
+		goto confbridge_cleanup;
+	}
+
+	leave_hook_data = ast_malloc(sizeof(*leave_hook_data));
+	if (!leave_hook_data) {
+		/* join_hook_data is cleaned up by ast_bridge_features_cleanup via the goto */
+		res = -1;
+		goto confbridge_cleanup;
+	}
+	leave_hook_data->user = &user;
+	leave_hook_data->conference = conference;
+	leave_hook_data->hook_type = AST_BRIDGE_HOOK_TYPE_LEAVE;
+	res = ast_bridge_leave_hook(&user.features, send_event_hook_callback,
+		leave_hook_data, ast_free_ptr, 0);
+	if (res) {
+		/* join_hook_data is cleaned up by ast_bridge_features_cleanup via the goto */
+		ast_free(leave_hook_data);
+		ast_log(LOG_ERROR, "Couldn't add bridge leave hook for channel '%s'\n", ast_channel_name(chan));
+		goto confbridge_cleanup;
+	}
 
 	if (ast_bridge_join_hook(&user.features, join_callback, NULL, NULL, 0)) {
 		async_play_sound_ready(user.chan);
@@ -2533,8 +2900,6 @@ static int confbridge_exec(struct ast_channel *chan, const char *data)
 		pbx_builtin_setvar_helper(chan, "CONFBRIDGE_RESULT", "HANGUP");
 	}
 
-	send_leave_event(&user, conference);
-
 	/* if we're shutting down, don't attempt to do further processing */
 	if (ast_shutting_down()) {
 		/*
@@ -3225,7 +3590,7 @@ static char *handle_cli_confbridge_list(struct ast_cli_entry *e, int cmd, struct
 	return CLI_SHOWUSAGE;
 }
 
-/* \internal
+/*! \internal
  * \brief finds a conference by name and locks/unlocks.
  *
  * \retval 0 success
@@ -3249,7 +3614,7 @@ static int generic_lock_unlock_helper(int lock, const char *conference_name)
 	return res;
 }
 
-/* \internal
+/*! \internal
  * \brief finds a conference user by channel name and mutes/unmutes them.
  *
  * \retval 0 success
@@ -3549,6 +3914,90 @@ static struct ast_custom_function confbridge_info_function = {
 	.read = func_confbridge_info,
 };
 
+static int func_confbridge_channels(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
+{
+	char *parse, *outbuf;
+	struct confbridge_conference *conference;
+	struct confbridge_user *user;
+	int bytes, count = 0;
+	size_t outlen;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(type);
+		AST_APP_ARG(confno);
+	);
+
+	/* parse all the required arguments and make sure they exist. */
+	if (ast_strlen_zero(data)) {
+		return -1;
+	}
+	parse = ast_strdupa(data);
+	AST_STANDARD_APP_ARGS(args, parse);
+	if (ast_strlen_zero(args.confno) || ast_strlen_zero(args.type)) {
+		ast_log(LOG_WARNING, "Usage: %s(category,confno)", cmd);
+		return -1;
+	}
+	conference = ao2_find(conference_bridges, args.confno, OBJ_KEY);
+	if (!conference) {
+		ast_debug(1, "No such conference: %s\n", args.confno);
+		return -1;
+	}
+
+	outbuf = buf;
+	outlen = len;
+
+	ao2_lock(conference);
+	if (!strcasecmp(args.type, "parties")) {
+		AST_LIST_TRAVERSE(&conference->active_list, user, list) {
+			bytes = snprintf(outbuf, outlen, "%s%s", count++ ? "," : "", ast_channel_name(user->chan));
+			outbuf += bytes;
+			outlen -= bytes;
+		}
+		AST_LIST_TRAVERSE(&conference->waiting_list, user, list) {
+			bytes = snprintf(outbuf, outlen, "%s%s", count++ ? "," : "", ast_channel_name(user->chan));
+			outbuf += bytes;
+			outlen -= bytes;
+		}
+	} else if (!strcasecmp(args.type, "active")) {
+		AST_LIST_TRAVERSE(&conference->active_list, user, list) {
+			bytes = snprintf(outbuf, outlen, "%s%s", count++ ? "," : "", ast_channel_name(user->chan));
+			outbuf += bytes;
+			outlen -= bytes;
+		}
+	} else if (!strcasecmp(args.type, "waiting")) {
+		AST_LIST_TRAVERSE(&conference->waiting_list, user, list) {
+			bytes = snprintf(outbuf, outlen, "%s%s", count++ ? "," : "", ast_channel_name(user->chan));
+			outbuf += bytes;
+			outlen -= bytes;
+		}
+	} else if (!strcasecmp(args.type, "admins")) {
+		AST_LIST_TRAVERSE(&conference->active_list, user, list) {
+			if (ast_test_flag(&user->u_profile, USER_OPT_ADMIN)) {
+				bytes = snprintf(outbuf, outlen, "%s%s", count++ ? "," : "", ast_channel_name(user->chan));
+				outbuf += bytes;
+				outlen -= bytes;
+			}
+		}
+	} else if (!strcasecmp(args.type, "marked")) {
+		AST_LIST_TRAVERSE(&conference->active_list, user, list) {
+			if (ast_test_flag(&user->u_profile, USER_OPT_MARKEDUSER)) {
+				bytes = snprintf(outbuf, outlen, "%s%s", count++ ? "," : "", ast_channel_name(user->chan));
+				outbuf += bytes;
+				outlen -= bytes;
+			}
+		}
+	} else {
+		ast_log(LOG_ERROR, "Invalid keyword '%s' passed to %s.\n", args.type, cmd);
+	}
+	ao2_unlock(conference);
+	ao2_ref(conference, -1);
+	return 0;
+}
+
+static struct ast_custom_function confbridge_channels_function = {
+	.name = "CONFBRIDGE_CHANNELS",
+	.read = func_confbridge_channels,
+};
+
 static int action_confbridgelist_item(struct mansession *s, const char *id_text, struct confbridge_conference *conference, struct confbridge_user *user, int waiting)
 {
 	struct ast_channel_snapshot *snapshot;
@@ -3573,6 +4022,7 @@ static int action_confbridgelist_item(struct mansession *s, const char *id_text,
 		"MarkedUser: %s\r\n"
 		"WaitMarked: %s\r\n"
 		"EndMarked: %s\r\n"
+		"EndMarkedAny: %s\r\n"
 		"Waiting: %s\r\n"
 		"Muted: %s\r\n"
 		"Talking: %s\r\n"
@@ -3585,6 +4035,7 @@ static int action_confbridgelist_item(struct mansession *s, const char *id_text,
 		AST_YESNO(ast_test_flag(&user->u_profile, USER_OPT_MARKEDUSER)),
 		AST_YESNO(ast_test_flag(&user->u_profile, USER_OPT_WAITMARKED)),
 		AST_YESNO(ast_test_flag(&user->u_profile, USER_OPT_ENDMARKED)),
+		AST_YESNO(ast_test_flag(&user->u_profile, USER_OPT_ENDMARKEDANY)),
 		AST_YESNO(waiting),
 		AST_YESNO(user->muted),
 		AST_YESNO(user->talking),
@@ -3984,6 +4435,53 @@ static int func_confbridge_info(struct ast_channel *chan, const char *cmd, char
 	return 0;
 }
 
+static int confkick_exec(struct ast_channel *chan, const char *data)
+{
+	char *parse;
+	struct confbridge_conference *conference;
+	int not_found;
+
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(confbridge);
+		AST_APP_ARG(channel);
+	);
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "No conference bridge specified.\n");
+		pbx_builtin_setvar_helper(chan, "CONFKICKSTATUS", "FAILURE");
+		return 0;
+	}
+
+	parse = ast_strdupa(data);
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	conference = ao2_find(conference_bridges, args.confbridge, OBJ_KEY);
+	if (!conference) {
+		ast_log(LOG_WARNING, "No conference bridge named '%s' found!\n", args.confbridge);
+		pbx_builtin_setvar_helper(chan, "CONFKICKSTATUS", "FAILURE");
+		return 0;
+	}
+	if (ast_strlen_zero(args.channel)) {
+		not_found = kick_conference_participant(conference, "all");
+	} else {
+		not_found = kick_conference_participant(conference, args.channel);
+	}
+
+	ao2_ref(conference, -1);
+	if (not_found) {
+		if (ast_strlen_zero(args.channel) || !strcasecmp("all", args.channel) || !strcasecmp("participants", args.channel)) {
+			ast_log(LOG_WARNING, "No participants found in conference bridge '%s'!\n", args.confbridge);
+		} else {
+			ast_log(LOG_WARNING, "No participant named '%s' found in conference bridge '%s'!\n", args.channel, args.confbridge);
+		}
+		pbx_builtin_setvar_helper(chan, "CONFKICKSTATUS", "FAILURE");
+		return 0;
+	}
+	ast_debug(1, "Kicked '%s' out of conference '%s'\n", args.channel, args.confbridge);
+	pbx_builtin_setvar_helper(chan, "CONFKICKSTATUS", "SUCCESS");
+	return 0;
+}
+
 void conf_add_user_active(struct confbridge_conference *conference, struct confbridge_user *user)
 {
 	AST_LIST_INSERT_TAIL(&conference->active_list, user, list);
@@ -4039,8 +4537,6 @@ void conf_remove_user_waiting(struct confbridge_conference *conference, struct c
  * \since 12.0.0
  *
  * \param tech What to unregister.
- *
- * \return Nothing
  */
 static void unregister_channel_tech(struct ast_channel_tech *tech)
 {
@@ -4077,9 +4573,11 @@ static int register_channel_tech(struct ast_channel_tech *tech)
 static int unload_module(void)
 {
 	ast_unregister_application(app);
+	ast_unregister_application(app2);
 
 	ast_custom_function_unregister(&confbridge_function);
 	ast_custom_function_unregister(&confbridge_info_function);
+	ast_custom_function_unregister(&confbridge_channels_function);
 
 	ast_cli_unregister_multiple(cli_confbridge, ARRAY_LEN(cli_confbridge));
 
@@ -4135,8 +4633,9 @@ static int load_module(void)
 	}
 
 	/* Create a container to hold the conference bridges */
-	conference_bridges = ao2_container_alloc(CONFERENCE_BRIDGE_BUCKETS,
-		conference_bridge_hash_cb, conference_bridge_cmp_cb);
+	conference_bridges = ao2_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0,
+		CONFERENCE_BRIDGE_BUCKETS,
+		conference_bridge_hash_cb, NULL, conference_bridge_cmp_cb);
 	if (!conference_bridges) {
 		unload_module();
 		return AST_MODULE_LOAD_DECLINE;
@@ -4146,9 +4645,11 @@ static int load_module(void)
 	res |= manager_confbridge_init();
 
 	res |= ast_register_application_xml(app, confbridge_exec);
+	res |= ast_register_application_xml(app2, confkick_exec);
 
 	res |= ast_custom_function_register_escalating(&confbridge_function, AST_CFE_WRITE);
 	res |= ast_custom_function_register(&confbridge_info_function);
+	res |= ast_custom_function_register(&confbridge_channels_function);
 
 	res |= ast_cli_register_multiple(cli_confbridge, ARRAY_LEN(cli_confbridge));
 

apps/app_dahdiras.c

@@ -28,6 +28,8 @@
 /*** MODULEINFO
 	<depend>dahdi</depend>
 	<support_level>deprecated</support_level>
+	<deprecated_in>16</deprecated_in>
+	<removed_in>19</removed_in>
  ***/
 
 #include "asterisk.h"
@@ -113,8 +115,10 @@ static pid_t spawn_ras(struct ast_channel *chan, char *args)
 		c = strsep(&stringp, ",");
 	}
 
-	argv[argc++] = "plugin";
-	argv[argc++] = "dahdi.so";
+	if (geteuid() == 0) {
+		argv[argc++] = "plugin";
+		argv[argc++] = "dahdi.so";
+	}
 	argv[argc++] = "stdin";
 
 	/* Finally launch PPP */

apps/app_dial.c

@@ -74,7 +74,7 @@
 			Attempt to connect to another device or endpoint and bridge the call.
 		</synopsis>
 		<syntax>
-			<parameter name="Technology/Resource" required="true" argsep="&amp;">
+			<parameter name="Technology/Resource" required="false" argsep="&amp;">
 				<argument name="Technology/Resource" required="true">
 					<para>Specification of the device(s) to dial.  These must be in the format of
 					<literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
@@ -86,6 +86,7 @@
 					<para>If you need more than one enter them as
 					Technology2/Resource2&amp;Technology3/Resource3&amp;.....</para>
 				</argument>
+				<xi:include xpointer="xpointer(/docs/info[@name='Dial_Resource'])" />
 			</parameter>
 			<parameter name="timeout" required="false">
 				<para>Specifies the number of seconds we attempt to dial the specified devices.</para>
@@ -93,11 +94,17 @@
 			</parameter>
 			<parameter name="options" required="false">
 				<optionlist>
-				<option name="A">
-					<argument name="x" required="true">
+				<option name="A" argsep=":">
+					<argument name="x">
 						<para>The file to play to the called party</para>
 					</argument>
-					<para>Play an announcement to the called party, where <replaceable>x</replaceable> is the prompt to be played</para>
+					<argument name="y">
+						<para>The file to play to the calling party</para>
+					</argument>
+					<para>Play an announcement to the called and/or calling parties, where <replaceable>x</replaceable>
+					is the prompt to be played to the called party and <replaceable>y</replaceable> is the prompt
+					to be played to the caller. The files may be different and will be played to each party
+					simultaneously.</para>
 				</option>
 				<option name="a">
 					<para>Immediately answer the calling channel when the called channel answers in
@@ -142,16 +149,18 @@
 					a call to be answered. Exit to that extension if it exists in the
 					current context, or the context defined in the <variable>EXITCONTEXT</variable> variable,
 					if it exists.</para>
-					<note>
-						<para>Many SIP and ISDN phones cannot send DTMF digits until the call is
-						connected.  If you wish to use this option with these phones, you
-						can use the <literal>Answer</literal> application before dialing.</para>
-					</note>
+					<para>NOTE: Many SIP and ISDN phones cannot send DTMF digits until the call is
+					connected.  If you wish to use this option with these phones, you
+					can use the <literal>Answer</literal> application before dialing.</para>
 				</option>
 				<option name="D" argsep=":">
 					<argument name="called" />
 					<argument name="calling" />
 					<argument name="progress" />
+					<argument name="mfprogress" />
+					<argument name="mfwink" />
+					<argument name="sfprogress" />
+					<argument name="sfwink" />
 					<para>Send the specified DTMF strings <emphasis>after</emphasis> the called
 					party has answered, but before the call gets bridged.  The
 					<replaceable>called</replaceable> DTMF string is sent to the called party, and the
@@ -159,6 +168,20 @@
 					can be used alone.  If <replaceable>progress</replaceable> is specified, its DTMF is sent
 					to the called party immediately after receiving a <literal>PROGRESS</literal> message.</para>
 					<para>See <literal>SendDTMF</literal> for valid digits.</para>
+					<para>If <replaceable>mfprogress</replaceable> is specified, its MF is sent
+					to the called party immediately after receiving a <literal>PROGRESS</literal> message.
+					If <replaceable>mfwink</replaceable> is specified, its MF is sent
+					to the called party immediately after receiving a <literal>WINK</literal> message.</para>
+					<para>See <literal>SendMF</literal> for valid digits.</para>
+					<para>If <replaceable>sfprogress</replaceable> is specified, its SF is sent
+					to the called party immediately after receiving a <literal>PROGRESS</literal> message.
+					If <replaceable>sfwink</replaceable> is specified, its SF is sent
+					to the called party immediately after receiving a <literal>WINK</literal> message.</para>
+					<para>See <literal>SendSF</literal> for valid digits.</para>
+				</option>
+				<option name="E">
+					<para>Enable echoing of sent MF or SF digits back to caller (e.g. "hearpulsing").
+					Used in conjunction with the D option.</para>
 				</option>
 				<option name="e">
 					<para>Execute the <literal>h</literal> extension for peer after the call ends</para>
@@ -177,21 +200,15 @@
 					<argument name="priority" required="true" />
 					<para>When the caller hangs up, transfer the <emphasis>called</emphasis> party
 					to the specified destination and <emphasis>start</emphasis> execution at that location.</para>
-					<note>
-						<para>Any channel variables you want the called channel to inherit from the caller channel must be
-						prefixed with one or two underbars ('_').</para>
-					</note>
+					<para>NOTE: Any channel variables you want the called channel to inherit from the caller channel must be
+					prefixed with one or two underbars ('_').</para>
 				</option>
 				<option name="F">
 					<para>When the caller hangs up, transfer the <emphasis>called</emphasis> party to the next priority of the current extension
 					and <emphasis>start</emphasis> execution at that location.</para>
-					<note>
-						<para>Any channel variables you want the called channel to inherit from the caller channel must be
-						prefixed with one or two underbars ('_').</para>
-					</note>
-					<note>
-						<para>Using this option from a Macro() or GoSub() might not make sense as there would be no return points.</para>
-					</note>
+					<para>NOTE: Any channel variables you want the called channel to inherit from the caller channel must be
+					prefixed with one or two underbars ('_').</para>
+					<para>NOTE: Using this option from a Macro() or GoSub() might not make sense as there would be no return points.</para>
 				</option>
 				<option name="g">
 					<para>Proceed with dialplan execution at the next priority in the current extension if the
@@ -204,9 +221,7 @@
 					<para>If the call is answered, transfer the calling party to
 					the specified <replaceable>priority</replaceable> and the called party to the specified
 					<replaceable>priority</replaceable> plus one.</para>
-					<note>
-						<para>You cannot use any additional action post answer options in conjunction with this option.</para>
-					</note>
+					<para>NOTE: You cannot use any additional action post answer options in conjunction with this option.</para>
 				</option>
 				<option name="h">
 					<para>Allow the called party to hang up by sending the DTMF sequence
@@ -215,12 +230,10 @@
 				<option name="H">
 					<para>Allow the calling party to hang up by sending the DTMF sequence
 					defined for disconnect in <filename>features.conf</filename>.</para>
-					<note>
-						<para>Many SIP and ISDN phones cannot send DTMF digits until the call is
-						connected.  If you wish to allow DTMF disconnect before the dialed
-						party answers with these phones, you can use the <literal>Answer</literal>
-						application before dialing.</para>
-					</note>
+					<para>NOTE: Many SIP and ISDN phones cannot send DTMF digits until the call is
+					connected.  If you wish to allow DTMF disconnect before the dialed
+					party answers with these phones, you can use the <literal>Answer</literal>
+					application before dialing.</para>
 				</option>
 				<option name="i">
 					<para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
@@ -316,18 +329,14 @@
 							</value>
 						</variable>
 					</variablelist>
-					<note>
-						<para>You cannot use any additional action post answer options in conjunction
-						with this option. Also, pbx services are run on the peer (called) channel,
-						so you will not be able to set timeouts via the <literal>TIMEOUT()</literal> function in this macro.</para>
-					</note>
-					<warning><para>Be aware of the limitations that macros have, specifically with regards to use of
+					<para>NOTE: You cannot use any additional action post answer options in conjunction
+					with this option. Also, pbx services are run on the peer (called) channel,
+					so you will not be able to set timeouts via the <literal>TIMEOUT()</literal> function in this macro.</para>
+					<para>WARNING: Be aware of the limitations that macros have, specifically with regards to use of
 					the <literal>WaitExten</literal> application. For more information, see the documentation for
-					<literal>Macro()</literal>.</para></warning>
-					<note>
-						<para>Macros are deprecated, GoSub should be used instead,
-						see the <literal>U</literal> option.</para>
-					</note>
+					<literal>Macro()</literal>.</para>
+					<para>NOTE: Macros are deprecated, GoSub should be used instead,
+					see the <literal>U</literal> option.</para>
 				</option>
 				<option name="n">
 					<argument name="delete">
@@ -363,7 +372,7 @@
 					</argument>
 					<para>Enables <emphasis>operator services</emphasis> mode.  This option only
 					works when bridging a DAHDI channel to another DAHDI channel
-					only. if specified on non-DAHDI interfaces, it will be ignored.
+					only. If specified on non-DAHDI interfaces, it will be ignored.
 					When the destination answers (presumably an operator services
 					station), the originator no longer has control of their line.
 					They may hang up, but the switch will not release their line
@@ -392,10 +401,8 @@
 						to send no cause.  See the <filename>causes.h</filename> file for the
 						full list of valid causes and names.
 						</para>
-					<note>
-						<para>chan_sip does not support setting the cause on a CANCEL to anything
-						other than ANSWERED_ELSEWHERE.</para>
-					</note>
+					<para>NOTE: chan_sip does not support setting the cause on a CANCEL to anything
+					other than ANSWERED_ELSEWHERE.</para>
 				</option>
 				<option name="r">
 					<para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
@@ -430,7 +437,8 @@
 				</option>
 				<option name="U" argsep="^">
 					<argument name="x" required="true">
-						<para>Name of the subroutine to execute via <literal>Gosub</literal></para>
+						<para>Name of the subroutine context to execute via <literal>Gosub</literal>.
+						The subroutine execution starts in the named context at the s exten and priority 1.</para>
 					</argument>
 					<argument name="arg" multiple="true" required="false">
 						<para>Arguments for the <literal>Gosub</literal> routine</para>
@@ -459,11 +467,9 @@
 							</value>
 						</variable>
 					</variablelist>
-					<note>
-						<para>You cannot use any additional action post answer options in conjunction
-						with this option. Also, pbx services are run on the peer (called) channel,
-						so you will not be able to set timeouts via the <literal>TIMEOUT()</literal> function in this routine.</para>
-					</note>
+					<para>NOTE: You cannot use any additional action post answer options in conjunction
+					with this option. Also, pbx services are run on the <emphasis>called</emphasis> channel,
+					so you will not be able to set timeouts via the <literal>TIMEOUT()</literal> function in this routine.</para>
 				</option>
 				<option name="u">
 					<argument name = "x" required="true">
@@ -512,7 +518,6 @@
 			answered, if it has not already been answered. These two channels will then
 			be active in a bridged call. All other channels that were requested will then
 			be hung up.</para>
-
 			<para>Unless there is a timeout specified, the Dial application will wait
 			indefinitely until one of the called channels answers, the user hangs up, or
 			if all of the called channels are busy or unavailable. Dialplan execution will
@@ -525,7 +530,6 @@
 			If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
 			application will be put into that group (as in <literal>Set(GROUP()=...</literal>). Unlike <variable>OUTBOUND_GROUP</variable>,
 			however, the variable will be unset after use.</para>
-
 			<example title="Dial with 30 second timeout">
 			 same => n,Dial(PJSIP/alice,30)
 			</example>
@@ -547,28 +551,24 @@
 			</example>
 			<example title="Dial with pre-dial subroutines">
 			[default]
-
-			exten => callee_channel,1,NoOp()
+			exten => callee_channel,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
 			 same => n,Log(NOTICE, I'm called on channel ${CHANNEL} prior to it starting the dial attempt)
 			 same => n,Return()
-
-			exten => called_channel,1,NoOp()
+			exten => called_channel,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
 			 same => n,Log(NOTICE, I'm called on outbound channel ${CHANNEL} prior to it being used to dial someone)
 			 same => n,Return()
-
 			exten => _X.,1,NoOp()
-			 same => n,Dial(PJSIP/alice,,b(default^called_channel^1)B(default^callee_channel^1))
+			 same => n,Dial(PJSIP/alice,,b(default^called_channel^1(my_gosub_arg1^my_gosub_arg2))B(default^callee_channel^1(my_gosub_arg1^my_gosub_arg2)))
 			 same => n,Hangup()
 			</example>
 			<example title="Dial with post-answer subroutine executed on outbound channel">
-			[default]
-
-			exten => called_channel,1,NoOp()
+			[my_gosub_routine]
+			exten => s,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
 			 same => n,Playback(hello)
 			 same => n,Return()
-
+			[default]
 			exten => _X.,1,NoOp()
-			 same => n,Dial(PJSIP/alice,,U(default^called_channel^1))
+			 same => n,Dial(PJSIP/alice,,U(my_gosub_routine^my_gosub_arg1^my_gosub_arg2))
 			 same => n,Hangup()
 			</example>
 			<example title="Dial into ConfBridge using 'G' option">
@@ -582,9 +582,27 @@
 				<variable name="DIALEDTIME">
 					<para>This is the time from dialing a channel until when it is disconnected.</para>
 				</variable>
+				<variable name="DIALEDTIME_MS">
+					<para>This is the milliseconds version of the DIALEDTIME variable.</para>
+				</variable>
 				<variable name="ANSWEREDTIME">
 					<para>This is the amount of time for actual call.</para>
 				</variable>
+				<variable name="ANSWEREDTIME_MS">
+					<para>This is the milliseconds version of the ANSWEREDTIME variable.</para>
+				</variable>
+				<variable name="RINGTIME">
+					<para>This is the time from creating the channel to the first RINGING event received. Empty if there was no ring.</para>
+				</variable>
+				<variable name="RINGTIME_MS">
+					<para>This is the milliseconds version of the RINGTIME variable.</para>
+				</variable>
+				<variable name="PROGRESSTIME">
+					<para>This is the time from creating the channel to the first PROGRESS event received. Empty if there was no such event.</para>
+				</variable>
+				<variable name="PROGRESSTIME_MS">
+					<para>This is the milliseconds version of the PROGRESSTIME variable.</para>
+				</variable>
 				<variable name="DIALEDPEERNAME">
 					<para>The name of the outbound channel that answered the call.</para>
 				</variable>
@@ -596,12 +614,31 @@
 				</variable>
 				<variable name="DIALSTATUS">
 					<para>This is the status of the call</para>
-					<value name="CHANUNAVAIL" />
-					<value name="CONGESTION" />
-					<value name="NOANSWER" />
-					<value name="BUSY" />
-					<value name="ANSWER" />
-					<value name="CANCEL" />
+					<value name="CHANUNAVAIL">
+						Either the dialed peer exists but is not currently reachable, e.g.
+						endpoint is not registered, or an attempt was made to call a
+						nonexistent location, e.g. nonexistent DNS hostname.
+					</value>
+					<value name="CONGESTION">
+						Channel or switching congestion occured when routing the call.
+						This can occur if there is a slow or no response from the remote end.
+					</value>
+					<value name="NOANSWER">
+						Called party did not answer.
+					</value>
+					<value name="BUSY">
+						The called party was busy or indicated a busy status.
+						Note that some SIP devices will respond with 486 Busy if their Do Not Disturb
+						modes are active. In this case, you can use DEVICE_STATUS to check if the
+						endpoint is actually in use, if needed.
+					</value>
+					<value name="ANSWER">
+						The call was answered.
+						Any other result implicitly indicates the call was not answered.
+					</value>
+					<value name="CANCEL">
+						Dial was cancelled before call was answered or reached some other terminating event.
+					</value>
 					<value name="DONTCALL">
 						For the Privacy and Screening Modes.
 						Will be set if the called party chooses to send the calling party to the 'Go Away' script.
@@ -610,7 +647,9 @@
 						For the Privacy and Screening Modes.
 						Will be set if the called party chooses to send the calling party to the 'torture' script.
 					</value>
-					<value name="INVALIDARGS" />
+					<value name="INVALIDARGS">
+						Dial failed due to invalid syntax.
+					</value>
 				</variable>
 			</variablelist>
 		</description>
@@ -710,6 +749,7 @@ enum {
 #define OPT_PREDIAL_CALLER   (1LLU << 42)
 #define OPT_RING_WITH_EARLY_MEDIA (1LLU << 43)
 #define OPT_HANGUPCAUSE      (1LLU << 44)
+#define OPT_HEARPULSING      (1LLU << 45)
 
 enum {
 	OPT_ARG_ANNOUNCE = 0,
@@ -745,6 +785,7 @@ AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
 	AST_APP_OPTION('c', OPT_CANCEL_ELSEWHERE),
 	AST_APP_OPTION('d', OPT_DTMF_EXIT),
 	AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
+	AST_APP_OPTION('E', OPT_HEARPULSING),
 	AST_APP_OPTION('e', OPT_PEER_H),
 	AST_APP_OPTION_ARG('f', OPT_FORCECLID, OPT_ARG_FORCECLID),
 	AST_APP_OPTION_ARG('F', OPT_CALLEE_GO_ON, OPT_ARG_CALLEE_GO_ON),
@@ -923,7 +964,7 @@ static const char *get_cid_name(char *name, int namelen, struct ast_channel *cha
  * XXX this code is highly suspicious, as it essentially overwrites
  * the outgoing channel without properly deleting it.
  *
- * \todo eventually this function should be intergrated into and replaced by ast_call_forward()
+ * \todo eventually this function should be integrated into and replaced by ast_call_forward()
  */
 static void do_forward(struct chanlist *o, struct cause_args *num,
 	struct ast_flags64 *peerflags, int single, int caller_entertained, int *to,
@@ -1002,7 +1043,8 @@ static void do_forward(struct chanlist *o, struct cause_args *num,
 			 * any Dial operations that happen later won't record CC interfaces.
 			 */
 			ast_ignore_cc(o->chan);
-			ast_log(LOG_NOTICE, "Not accepting call completion offers from call-forward recipient %s\n", ast_channel_name(o->chan));
+			ast_verb(3, "Not accepting call completion offers from call-forward recipient %s\n",
+				ast_channel_name(o->chan));
 		} else
 			ast_log(LOG_NOTICE,
 				"Forwarding failed to create channel to dial '%s/%s' (cause = %d)\n",
@@ -1138,6 +1180,7 @@ struct privacy_args {
 	char privcid[256];
 	char privintro[1024];
 	char status[256];
+	int canceled;
 };
 
 static void publish_dial_end_event(struct ast_channel *in, struct dial_head *out_chans, struct ast_channel *exception, const char *status)
@@ -1159,8 +1202,6 @@ static void publish_dial_end_event(struct ast_channel *in, struct dial_head *out
  * \param chan Channel to get connected line updated.
  * \param peer Channel providing connected line information.
  * \param is_caller Non-zero if chan is the calling channel.
- *
- * \return Nothing
  */
 static void update_connected_line_from_peer(struct ast_channel *chan, struct ast_channel *peer, int is_caller)
 {
@@ -1179,11 +1220,31 @@ static void update_connected_line_from_peer(struct ast_channel *chan, struct ast
 	ast_party_connected_line_free(&connected_caller);
 }
 
+/*!
+ * \internal
+ * \pre chan is locked
+ */
+static void set_duration_var(struct ast_channel *chan, const char *var_base, int64_t duration)
+{
+	char buf[32];
+	char full_var_name[128];
+
+	snprintf(buf, sizeof(buf), "%" PRId64, duration / 1000);
+	pbx_builtin_setvar_helper(chan, var_base, buf);
+
+	snprintf(full_var_name, sizeof(full_var_name), "%s_MS", var_base);
+	snprintf(buf, sizeof(buf), "%" PRId64, duration);
+	pbx_builtin_setvar_helper(chan, full_var_name, buf);
+}
+
 static struct ast_channel *wait_for_answer(struct ast_channel *in,
 	struct dial_head *out_chans, int *to, struct ast_flags64 *peerflags,
 	char *opt_args[],
 	struct privacy_args *pa,
 	const struct cause_args *num_in, int *result, char *dtmf_progress,
+	char *mf_progress, char *mf_wink,
+	char *sf_progress, char *sf_wink,
+	const int hearpulsing,
 	const int ignore_cc,
 	struct ast_party_id *forced_clid, struct ast_party_id *stored_clid)
 {
@@ -1201,6 +1262,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 	int is_cc_recall;
 	int cc_frame_received = 0;
 	int num_ringing = 0;
+	int sent_ring = 0;
+	int sent_progress = 0, sent_wink = 0;
 	struct timeval start = ast_tvnow();
 
 	if (single) {
@@ -1474,6 +1537,23 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 							ast_indicate(in, AST_CONTROL_RINGING);
 							pa->sentringing++;
 						}
+						if (!sent_ring) {
+							struct timeval now, then;
+							int64_t diff;
+
+							now = ast_tvnow();
+
+							ast_channel_lock(in);
+							ast_channel_stage_snapshot(in);
+
+							then = ast_channel_creationtime(c);
+							diff = ast_tvzero(then) ? 0 : ast_tvdiff_ms(now, then);
+							set_duration_var(in, "RINGTIME", diff);
+
+							ast_channel_stage_snapshot_done(in);
+							ast_channel_unlock(in);
+							sent_ring = 1;
+						}
 					}
 					ast_channel_publish_dial(in, c, NULL, "RINGING");
 					break;
@@ -1489,14 +1569,72 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 							ast_indicate(in, AST_CONTROL_PROGRESS);
 						}
 					}
-					if (!ast_strlen_zero(dtmf_progress)) {
-						ast_verb(3,
-							"Sending DTMF '%s' to the called party as result of receiving a PROGRESS message.\n",
-							dtmf_progress);
-						ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
+					if (!sent_progress) {
+						struct timeval now, then;
+						int64_t diff;
+
+						now = ast_tvnow();
+
+						ast_channel_lock(in);
+						ast_channel_stage_snapshot(in);
+
+						then = ast_channel_creationtime(c);
+						diff = ast_tvzero(then) ? 0 : ast_tvdiff_ms(now, then);
+						set_duration_var(in, "PROGRESSTIME", diff);
+
+						ast_channel_stage_snapshot_done(in);
+						ast_channel_unlock(in);
+						sent_progress = 1;
+
+						if (!ast_strlen_zero(mf_progress)) {
+							ast_verb(3,
+								"Sending MF '%s' to %s as result of "
+								"receiving a PROGRESS message.\n",
+								mf_progress, hearpulsing ? "parties" : "called party");
+							ast_mf_stream(c, (hearpulsing ? NULL : in),
+							(hearpulsing ? in : NULL), mf_progress, 50, 55, 120, 65, 0);
+						}
+						if (!ast_strlen_zero(sf_progress)) {
+							ast_verb(3,
+								"Sending SF '%s' to %s as result of "
+								"receiving a PROGRESS message.\n",
+								sf_progress, (hearpulsing ? "parties" : "called party"));
+							ast_sf_stream(c, (hearpulsing ? NULL : in),
+							(hearpulsing ? in : NULL), sf_progress, 0, 0);
+						}
+						if (!ast_strlen_zero(dtmf_progress)) {
+							ast_verb(3,
+								"Sending DTMF '%s' to the called party as result of "
+								"receiving a PROGRESS message.\n",
+								dtmf_progress);
+							ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
+						}
 					}
 					ast_channel_publish_dial(in, c, NULL, "PROGRESS");
 					break;
+				case AST_CONTROL_WINK:
+					ast_verb(3, "%s winked, passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
+					if (!sent_wink) {
+						sent_wink = 1;
+						if (!ast_strlen_zero(mf_wink)) {
+							ast_verb(3,
+								"Sending MF '%s' to %s as result of "
+								"receiving a WINK message.\n",
+								mf_wink, (hearpulsing ? "parties" : "called party"));
+							ast_mf_stream(c, (hearpulsing ? NULL : in),
+							(hearpulsing ? in : NULL), mf_wink, 50, 55, 120, 65, 0);
+						}
+						if (!ast_strlen_zero(sf_wink)) {
+							ast_verb(3,
+								"Sending SF '%s' to %s as result of "
+								"receiving a WINK message.\n",
+								sf_wink, (hearpulsing ? "parties" : "called party"));
+							ast_sf_stream(c, (hearpulsing ? NULL : in),
+							(hearpulsing ? in : NULL), sf_wink, 0, 0);
+						}
+					}
+					ast_indicate(in, AST_CONTROL_WINK);
+					break;
 				case AST_CONTROL_VIDUPDATE:
 				case AST_CONTROL_SRCUPDATE:
 				case AST_CONTROL_SRCCHANGE:
@@ -1641,6 +1779,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 				/* Got hung up */
 				*to = -1;
 				strcpy(pa->status, "CANCEL");
+				pa->canceled = 1;
 				publish_dial_end_event(in, out_chans, NULL, pa->status);
 				if (f) {
 					if (f->data.uint32) {
@@ -1665,6 +1804,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 						*to = 0;
 						*result = f->subclass.integer;
 						strcpy(pa->status, "CANCEL");
+						pa->canceled = 1;
 						publish_dial_end_event(in, out_chans, NULL, pa->status);
 						ast_frfree(f);
 						ast_channel_unlock(in);
@@ -1681,6 +1821,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 					ast_verb(3, "User requested call disconnect.\n");
 					*to = 0;
 					strcpy(pa->status, "CANCEL");
+					pa->canceled = 1;
 					publish_dial_end_event(in, out_chans, NULL, pa->status);
 					ast_frfree(f);
 					if (is_cc_recall) {
@@ -1734,6 +1875,10 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 						ast_verb(3, "Call on %s left from hold\n", ast_channel_name(o->chan));
 						ast_indicate(o->chan, AST_CONTROL_UNHOLD);
 						break;
+					case AST_CONTROL_FLASH:
+						ast_verb(3, "Hook flash on %s\n", ast_channel_name(o->chan));
+						ast_indicate(o->chan, AST_CONTROL_FLASH);
+						break;
 					case AST_CONTROL_VIDUPDATE:
 					case AST_CONTROL_SRCUPDATE:
 					case AST_CONTROL_SRCCHANGE:
@@ -1750,12 +1895,20 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 						ast_indicate(o->chan, f->subclass.integer);
 						break;
 					case AST_CONTROL_CONNECTED_LINE:
+						if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
+							ast_verb(3, "Connected line update to %s prevented.\n", ast_channel_name(o->chan));
+							break;
+						}
 						if (ast_channel_connected_line_sub(in, o->chan, f, 1) &&
 							ast_channel_connected_line_macro(in, o->chan, f, 0, 1)) {
 							ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
 						}
 						break;
 					case AST_CONTROL_REDIRECTING:
+						if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
+							ast_verb(3, "Redirecting update to %s prevented.\n", ast_channel_name(o->chan));
+							break;
+						}
 						if (ast_channel_redirecting_sub(in, o->chan, f, 1) &&
 							ast_channel_redirecting_macro(in, o->chan, f, 0, 1)) {
 							ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
@@ -1941,7 +2094,7 @@ static int do_privacy(struct ast_channel *chan, struct ast_channel *peer,
 		/* well, there seems basically two choices. Just patch the caller thru immediately,
 			  or,... put 'em thru to voicemail. */
 		/* since the callee may have hung up, let's do the voicemail thing, no database decision */
-		ast_log(LOG_NOTICE, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
+		ast_verb(3, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
 		/* XXX should we set status to DENY ? */
 		/* XXX what about the privacy flags ? */
 		break;
@@ -2040,9 +2193,7 @@ static int setup_privacy_args(struct privacy_args *pa,
 			/* the file doesn't exist yet. Let the caller submit his
 			   vocal intro for posterity */
 			/* priv-recordintro script:
-
 			   "At the tone, please say your name:"
-
 			*/
 			int silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
 			ast_answer(chan);
@@ -2067,18 +2218,12 @@ static int setup_privacy_args(struct privacy_args *pa,
 
 static void end_bridge_callback(void *data)
 {
-	char buf[80];
-	time_t end;
 	struct ast_channel *chan = data;
 
-	time(&end);
-
 	ast_channel_lock(chan);
 	ast_channel_stage_snapshot(chan);
-	snprintf(buf, sizeof(buf), "%d", ast_channel_get_up_time(chan));
-	pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
-	snprintf(buf, sizeof(buf), "%d", ast_channel_get_duration(chan));
-	pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
+	set_duration_var(chan, "ANSWEREDTIME", ast_channel_get_up_time_ms(chan));
+	set_duration_var(chan, "DIALEDTIME", ast_channel_get_duration_ms(chan));
 	ast_channel_stage_snapshot_done(chan);
 	ast_channel_unlock(chan);
 }
@@ -2126,8 +2271,6 @@ static int dial_handle_playtones(struct ast_channel *chan, const char *data)
  * \param peer Peer channel for bridge.
  * \param opts Dialing option flags.
  * \param opt_args Dialing option argument strings.
- *
- * \return Nothing
  */
 static void setup_peer_after_bridge_goto(struct ast_channel *chan, struct ast_channel *peer, struct ast_flags64 *opts, char *opt_args[])
 {
@@ -2161,15 +2304,18 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 	struct ast_channel *peer;
 	int to; /* timeout */
 	struct cause_args num = { chan, 0, 0, 0 };
-	int cause;
+	int cause, hanguptreecause = -1;
 
 	struct ast_bridge_config config = { { 0, } };
 	struct timeval calldurationlimit = { 0, };
-	char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress=NULL;
+	char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress = NULL;
+	char *mf_progress = NULL, *mf_wink = NULL;
+	char *sf_progress = NULL, *sf_wink = NULL;
 	struct privacy_args pa = {
 		.sentringing = 0,
 		.privdb_val = 0,
 		.status = "INVALIDARGS",
+		.canceled = 0,
 	};
 	int sentringing = 0, moh = 0;
 	const char *outbound_group = NULL;
@@ -2220,7 +2366,13 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 	pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
 	pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
 	pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
+	pbx_builtin_setvar_helper(chan, "ANSWEREDTIME_MS", "");
 	pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
+	pbx_builtin_setvar_helper(chan, "DIALEDTIME_MS", "");
+	pbx_builtin_setvar_helper(chan, "RINGTIME", "");
+	pbx_builtin_setvar_helper(chan, "RINGTIME_MS", "");
+	pbx_builtin_setvar_helper(chan, "PROGRESSTIME", "");
+	pbx_builtin_setvar_helper(chan, "PROGRESSTIME_MS", "");
 	ast_channel_stage_snapshot_done(chan);
 	max_forwards = ast_max_forwards_get(chan);
 	ast_channel_unlock(chan);
@@ -2232,12 +2384,6 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 		return -1;
 	}
 
-	if (ast_strlen_zero(data)) {
-		ast_log(LOG_WARNING, "Dial requires an argument (technology/resource)\n");
-		pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
-		return -1;
-	}
-
 	if (ast_check_hangup_locked(chan)) {
 		/*
 		 * Caller hung up before we could dial.  If dial is executed
@@ -2256,7 +2402,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 		return -1;
 	}
 
-	parse = ast_strdupa(data);
+	parse = ast_strdupa(data ?: "");
 
 	AST_STANDARD_APP_ARGS(args, parse);
 
@@ -2266,12 +2412,6 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 		goto done;
 	}
 
-	if (ast_strlen_zero(args.peers)) {
-		ast_log(LOG_WARNING, "Dial requires an argument (technology/resource)\n");
-		pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
-		goto done;
-	}
-
 	if (ast_cc_call_init(chan, &ignore_cc)) {
 		goto done;
 	}
@@ -2297,7 +2437,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 	if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
 		calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
 		if (!calldurationlimit.tv_sec) {
-			ast_log(LOG_WARNING, "Dial does not accept S(%s), hanging up.\n", opt_args[OPT_ARG_DURATION_STOP]);
+			ast_log(LOG_WARNING, "Dial does not accept S(%s)\n", opt_args[OPT_ARG_DURATION_STOP]);
 			pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
 			goto done;
 		}
@@ -2305,9 +2445,13 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 	}
 
 	if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
-		dtmf_progress = opt_args[OPT_ARG_SENDDTMF];
-		dtmfcalled = strsep(&dtmf_progress, ":");
-		dtmfcalling = strsep(&dtmf_progress, ":");
+		sf_wink = opt_args[OPT_ARG_SENDDTMF];
+		dtmfcalled = strsep(&sf_wink, ":");
+		dtmfcalling = strsep(&sf_wink, ":");
+		dtmf_progress = strsep(&sf_wink, ":");
+		mf_progress = strsep(&sf_wink, ":");
+		mf_wink = strsep(&sf_wink, ":");
+		sf_progress = strsep(&sf_wink, ":");
 	}
 
 	if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
@@ -2444,14 +2588,25 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 
 	/* loop through the list of dial destinations */
 	rest = args.peers;
-	while ((cur = strsep(&rest, "&")) ) {
+	while ((cur = strsep(&rest, "&"))) {
 		struct ast_channel *tc; /* channel for this destination */
-		/* Get a technology/resource pair */
-		char *number = cur;
-		char *tech = strsep(&number, "/");
+		char *number;
+		char *tech;
+		int i;
 		size_t tech_len;
 		size_t number_len;
 		struct ast_stream_topology *topology;
+		struct ast_stream *stream;
+
+		cur = ast_strip(cur);
+		if (ast_strlen_zero(cur)) {
+			/* No tech/resource in this position. */
+			continue;
+		}
+
+		/* Get a technology/resource pair */
+		number = cur;
+		tech = strsep(&number, "/");
 
 		num_dialed++;
 		if (ast_strlen_zero(number)) {
@@ -2507,13 +2662,29 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 
 		ast_channel_unlock(chan);
 
+		for (i = 0; i < ast_stream_topology_get_count(topology); ++i) {
+			stream = ast_stream_topology_get_stream(topology, i);
+			/* For both recvonly and sendonly the stream state reflects our state, that is we
+			 * are receiving only and we are sending only. Since we are requesting a
+			 * channel for the peer, we need to swap this to reflect what we will be doing.
+			 * That is, if we are receiving from Alice then we want to be sending to Bob,
+			 * so swap recvonly to sendonly and vice versa.
+			 */
+			if (ast_stream_get_state(stream) == AST_STREAM_STATE_RECVONLY) {
+				ast_stream_set_state(stream, AST_STREAM_STATE_SENDONLY);
+			} else if (ast_stream_get_state(stream) == AST_STREAM_STATE_SENDONLY) {
+				ast_stream_set_state(stream, AST_STREAM_STATE_RECVONLY);
+			}
+		}
+
 		tc = ast_request_with_stream_topology(tmp->tech, topology, NULL, chan, tmp->number, &cause);
 
 		ast_stream_topology_free(topology);
 
 		if (!tc) {
 			/* If we can't, just go on to the next call */
-			ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n",
+			/* Failure doesn't necessarily mean user error. DAHDI channels could be busy. */
+			ast_log(LOG_NOTICE, "Unable to create channel of type '%s' (cause %d - %s)\n",
 				tmp->tech, cause, ast_cause2str(cause));
 			handle_cause(cause, &num);
 			if (!rest) {
@@ -2651,6 +2822,19 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 		AST_LIST_INSERT_TAIL(&out_chans, tmp, node);
 	}
 
+	/* As long as we attempted to dial valid peers, don't throw a warning. */
+	/* If a DAHDI peer is busy, out_chans will be empty so checking list size is misleading. */
+	if (!num_dialed) {
+		ast_verb(3, "No devices or endpoints to dial (technology/resource)\n");
+		if (continue_exec) {
+			/* There is no point in having RetryDial try again */
+			*continue_exec = 1;
+		}
+		strcpy(pa.status, "CHANUNAVAIL");
+		res = 0;
+		goto out;
+	}
+
 	/*
 	 * PREDIAL: Run gosub on all of the callee channels
 	 *
@@ -2764,7 +2948,9 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 	}
 
 	peer = wait_for_answer(chan, &out_chans, &to, peerflags, opt_args, &pa, &num, &result,
-		dtmf_progress, ignore_cc, &forced_clid, &stored_clid);
+		dtmf_progress, mf_progress, mf_wink, sf_progress, sf_wink,
+		(ast_test_flag64(&opts, OPT_HEARPULSING) ? 1 : 0),
+		ignore_cc, &forced_clid, &stored_clid);
 
 	if (!peer) {
 		if (result) {
@@ -2848,33 +3034,71 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 			int digit = 0;
 			struct ast_channel *chans[2];
 			struct ast_channel *active_chan;
+			char *calledfile = NULL, *callerfile = NULL;
+			int calledstream = 0, callerstream = 0;
 
 			chans[0] = chan;
 			chans[1] = peer;
 
-			/* we need to stream the announcement while monitoring the caller for a hangup */
+			/* we need to stream the announcement(s) when the OPT_ARG_ANNOUNCE (-A) is set */
+			callerfile = opt_args[OPT_ARG_ANNOUNCE];
+			calledfile = strsep(&callerfile, ":");
 
-			/* stream the file */
-			res = ast_streamfile(peer, opt_args[OPT_ARG_ANNOUNCE], ast_channel_language(peer));
-			if (res) {
-				res = 0;
-				ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", opt_args[OPT_ARG_ANNOUNCE]);
+			/* stream the file(s) */
+			if (!ast_strlen_zero(calledfile)) {
+				res = ast_streamfile(peer, calledfile, ast_channel_language(peer));
+				if (res) {
+					res = 0;
+					ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", calledfile);
+				} else {
+					calledstream = 1;
+				}
+			}
+			if (!ast_strlen_zero(callerfile)) {
+				res = ast_streamfile(chan, callerfile, ast_channel_language(chan));
+				if (res) {
+					res = 0;
+					ast_log(LOG_ERROR, "error streaming file '%s' to caller\n", callerfile);
+				} else {
+					callerstream = 1;
+				}
 			}
 
+			/* can't use ast_waitstream, because we're streaming two files at once, and can't block
+				We'll need to handle both channels at once. */
+
 			ast_channel_set_flag(peer, AST_FLAG_END_DTMF_ONLY);
-			while (ast_channel_stream(peer)) {
-				int ms;
+			while (ast_channel_stream(peer) || ast_channel_stream(chan)) {
+				int mspeer, mschan;
 
-				ms = ast_sched_wait(ast_channel_sched(peer));
+				mspeer = ast_sched_wait(ast_channel_sched(peer));
+				mschan = ast_sched_wait(ast_channel_sched(chan));
 
-				if (ms < 0 && !ast_channel_timingfunc(peer)) {
-					ast_stopstream(peer);
+				if (calledstream) {
+					if (mspeer < 0 && !ast_channel_timingfunc(peer)) {
+						ast_stopstream(peer);
+						calledstream = 0;
+					}
+				}
+				if (callerstream) {
+					if (mschan < 0 && !ast_channel_timingfunc(chan)) {
+						ast_stopstream(chan);
+						callerstream = 0;
+					}
+				}
+
+				if (!calledstream && !callerstream) {
 					break;
 				}
-				if (ms < 0)
-					ms = 1000;
 
-				active_chan = ast_waitfor_n(chans, 2, &ms);
+				if (mspeer < 0)
+					mspeer = 1000;
+
+				if (mschan < 0)
+					mschan = 1000;
+
+				/* wait for the lowest maximum of the two */
+				active_chan = ast_waitfor_n(chans, 2, (mspeer > mschan ? &mschan : &mspeer));
 				if (active_chan) {
 					struct ast_channel *other_chan;
 					struct ast_frame *fr = ast_read(active_chan);
@@ -2924,6 +3148,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 					ast_frfree(fr);
 				}
 				ast_sched_runq(ast_channel_sched(peer));
+				ast_sched_runq(ast_channel_sched(chan));
 			}
 			ast_channel_clear_flag(peer, AST_FLAG_END_DTMF_ONLY);
 		}
@@ -3215,11 +3440,16 @@ out:
 	}
 
 	ast_channel_early_bridge(chan, NULL);
-	 /* forward 'answered elsewhere' if we received it */
-	hanguptree(&out_chans, NULL,
-		ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE
-		|| ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE)
-		? AST_CAUSE_ANSWERED_ELSEWHERE : -1);
+	/* forward 'answered elsewhere' if we received it */
+	if (ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE || ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE)) {
+		hanguptreecause = AST_CAUSE_ANSWERED_ELSEWHERE;
+	} else if (pa.canceled) { /* Caller canceled */
+		if (ast_channel_hangupcause(chan))
+			hanguptreecause = ast_channel_hangupcause(chan);
+		else
+			hanguptreecause = AST_CAUSE_NORMAL_CLEARING;
+	}
+	hanguptree(&out_chans, NULL, hanguptreecause);
 	pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
 	ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
 

apps/app_dictate.c

@@ -149,7 +149,8 @@ static int dictate_exec(struct ast_channel *chan, const char *data)
 		ast_mkdir(base, 0755);
 		len = strlen(base) + strlen(filein) + 2;
 		if (!path || len > maxlen) {
-			path = ast_alloca(len);
+			ast_free(path);
+			path = ast_malloc(len);
 			memset(path, 0, len);
 			maxlen = len;
 		} else {
@@ -334,6 +335,7 @@ static int dictate_exec(struct ast_channel *chan, const char *data)
 			ast_frfree(f);
 		}
 	}
+	ast_free(path);
 	if (oldr) {
 		ast_set_read_format(chan, oldr);
 		ao2_ref(oldr, -1);

apps/app_directory.c

@@ -396,7 +396,7 @@ static int select_item_menu(struct ast_channel *chan, struct directory_item **it
 {
 	struct directory_item **block, *item;
 	int i, limit, res = 0;
-	char buf[9];
+	char buf[7+12]; /* INT_MIN has a length of 12 chars */
 
 	/* option p(n): cellphone pause option */
 	select_item_pause(chan, flags, opts);

apps/app_disa.c

@@ -361,7 +361,6 @@ static int disa_exec(struct ast_channel *chan, const char *data)
 
 	if (k == 3) {
 		int recheck = 0;
-		struct ast_app *app_reset_cdr;
 
 		if (!ast_exists_extension(chan, args.context, exten, 1,
 			S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL))) {
@@ -386,10 +385,7 @@ static int disa_exec(struct ast_channel *chan, const char *data)
 				ast_channel_unlock(chan);
 			}
 
-			app_reset_cdr = pbx_findapp("ResetCDR");
-			if (app_reset_cdr) {
-				pbx_exec(chan, app_reset_cdr, special_noanswer ? "" : "e");
-			} else {
+			if (ast_pbx_exec_application(chan, "ResetCDR", special_noanswer ? "" : "e")) {
 				ast_log(AST_LOG_NOTICE, "ResetCDR application not found; CDR will not be reset\n");
 			}
 			ast_explicit_goto(chan, args.context, exten, 1);

apps/app_dtmfstore.c

@@ -0,0 +1,297 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2021, Naveen Albert
+ *
+ * Naveen Albert <asterisk@phreaknet.org>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Technology independent asynchronous DTMF collection
+ *
+ * \author Naveen Albert <asterisk@phreaknet.org>
+ *
+ * \ingroup functions
+ *
+ */
+
+/*** MODULEINFO
+	<support_level>extended</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include "asterisk/module.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/framehook.h"
+#include "asterisk/app.h"
+#include "asterisk/conversions.h"
+
+/*** DOCUMENTATION
+	<application name="StoreDTMF" language="en_US">
+		<since>
+			<version>16.20.0</version>
+			<version>18.6.0</version>
+			<version>19.0.0</version>
+		</since>
+		<synopsis>
+			Stores DTMF digits transmitted or received on a channel.
+		</synopsis>
+		<syntax>
+			<parameter name="direction" required="true">
+				<para>Must be <literal>TX</literal> or <literal>RX</literal> to
+				store digits, or <literal>remove</literal> to disable.</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>The StoreDTMF function can be used to obtain digits sent in the
+			<literal>TX</literal> or <literal>RX</literal> direction of any channel.</para>
+			<para>The arguments are:</para>
+			<para><replaceable>var_name</replaceable>: Name of variable to which to append
+			digits.</para>
+			<para><replaceable>max_digits</replaceable>: The maximum number of digits to
+			store in the variable. Defaults to 0 (no maximum). After reading <literal>
+			maximum</literal> digits, no more digits will be stored.</para>
+			<example title="Store digits in CDR variable">
+			same => n,StoreDTMF(TX,CDR(digits))
+			</example>
+			<example title="Store up to 24 digits">
+			same => n,StoreDTMF(RX,testvar,24)
+			</example>
+			<example title="Disable digit collection">
+			same => n,StoreDTMF(remove)
+			</example>
+		</description>
+	</application>
+ ***/
+
+static char *app = "StoreDTMF";
+
+/*! \brief Private data structure used with the function's datastore */
+struct dtmf_store_data {
+	int framehook_id;
+	char *rx_var;
+	char *tx_var;
+	int maxdigits;
+};
+
+static void datastore_destroy_cb(void *data) {
+	struct dtmf_store_data *d;
+	d = data;
+	if (d) {
+		if (d->rx_var) {
+			ast_free(d->rx_var);
+		}
+		if (d->tx_var) {
+			ast_free(d->tx_var);
+		}
+		ast_free(data);
+	}
+}
+
+/*! \brief The channel datastore the function uses to store state */
+static const struct ast_datastore_info dtmf_store_datastore = {
+	.type = "dtmf_store",
+	.destroy = datastore_destroy_cb
+};
+
+/*! \internal \brief Store digits tx/rx on the channel */
+static int remove_dtmf_store(struct ast_channel *chan)
+{
+	struct ast_datastore *datastore = NULL;
+	struct dtmf_store_data *data;
+	SCOPED_CHANNELLOCK(chan_lock, chan);
+
+	datastore = ast_channel_datastore_find(chan, &dtmf_store_datastore, NULL);
+	if (!datastore) {
+		ast_log(AST_LOG_WARNING, "Cannot remove StoreDTMF from %s: StoreDTMF not currently enabled\n",
+		        ast_channel_name(chan));
+		return -1;
+	}
+	data = datastore->data;
+
+	if (ast_framehook_detach(chan, data->framehook_id)) {
+		ast_log(AST_LOG_WARNING, "Failed to remove StoreDTMF framehook from channel %s\n",
+		        ast_channel_name(chan));
+		return -1;
+	}
+
+	if (ast_channel_datastore_remove(chan, datastore)) {
+		ast_log(AST_LOG_WARNING, "Failed to remove StoreDTMF datastore from channel %s\n",
+		        ast_channel_name(chan));
+		return -1;
+	}
+	ast_datastore_free(datastore);
+
+	return 0;
+}
+
+/*! \brief Frame hook that is called to intercept digit/undigit */
+static struct ast_frame *dtmf_store_framehook(struct ast_channel *chan,
+	struct ast_frame *f, enum ast_framehook_event event, void *data)
+{
+	char currentdata[512];
+	char varnamesub[64];
+	char *varname = NULL;
+	struct dtmf_store_data *framedata = data;
+	int len;
+
+	if (!f || !framedata) {
+		return f;
+	}
+
+	if ((event != AST_FRAMEHOOK_EVENT_WRITE) && (event != AST_FRAMEHOOK_EVENT_READ)) {
+		return f;
+	}
+
+	if (f->frametype != AST_FRAME_DTMF_END) {
+		return f;
+	}
+
+	/* If this is DTMF then store the digits */
+	if (event == AST_FRAMEHOOK_EVENT_READ && framedata->rx_var) { /* coming from source */
+		varname = framedata->rx_var;
+	} else if (event == AST_FRAMEHOOK_EVENT_WRITE && framedata->tx_var) { /* going to source */
+		varname = framedata->tx_var;
+	}
+
+	if (!varname) {
+		return f;
+	}
+
+	sprintf(varnamesub, "${%s}", varname);
+	pbx_substitute_variables_helper(chan, varnamesub, currentdata, 511);
+	/* pbx_builtin_getvar_helper works for regular vars but not CDR vars */
+	if (ast_strlen_zero(currentdata)) { /* var doesn't exist yet */
+		ast_debug(3, "Creating new digit store: %s\n", varname);
+	}
+	len = strlen(currentdata);
+	if (framedata->maxdigits > 0 && len >= framedata->maxdigits) {
+		ast_debug(3, "Reached digit limit: %d\n", framedata->maxdigits);
+		remove_dtmf_store(chan); /* reached max digit count, stop now */
+		return f;
+	} else {
+		char newdata[len + 2]; /* one more char + terminator */
+		if (len > 0) {
+			ast_copy_string(newdata, currentdata, len + 2);
+		}
+		newdata[len] = (unsigned) f->subclass.integer;
+		newdata[len + 1] = '\0';
+		ast_debug(3, "Appending to digit store: now %s\n", newdata);
+		pbx_builtin_setvar_helper(chan, varname, newdata);
+	}
+	return f;
+}
+
+/*! \internal \brief Enable digit interception on the channel */
+static int dtmfstore_exec(struct ast_channel *chan, const char *appdata)
+{
+	struct ast_datastore *datastore;
+	struct dtmf_store_data *data;
+	static struct ast_framehook_interface digit_framehook_interface = {
+		.version = AST_FRAMEHOOK_INTERFACE_VERSION,
+		.event_cb = dtmf_store_framehook,
+		.disable_inheritance = 1,
+	};
+	char *parse = ast_strdupa(appdata);
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(direction);
+		AST_APP_ARG(varname);
+		AST_APP_ARG(maxdigits);
+	);
+	SCOPED_CHANNELLOCK(chan_lock, chan);
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	if (ast_strlen_zero(appdata)) {
+		ast_log(AST_LOG_WARNING, "StoreDTMF requires an argument\n");
+		return -1;
+	}
+
+	if (!strcasecmp(args.direction, "remove")) {
+		return remove_dtmf_store(chan);
+	}
+
+	datastore = ast_channel_datastore_find(chan, &dtmf_store_datastore, NULL);
+	if (datastore) {
+		ast_log(AST_LOG_WARNING, "StoreDTMF already set on '%s'\n",
+		        ast_channel_name(chan));
+		return 0;
+	}
+
+	datastore = ast_datastore_alloc(&dtmf_store_datastore, NULL);
+	if (!datastore) {
+		return -1;
+	}
+
+	data = ast_calloc(1, sizeof(*data));
+	if (!data) {
+		ast_datastore_free(datastore);
+		return -1;
+	}
+
+	digit_framehook_interface.data = data;
+
+	data->rx_var = NULL;
+	data->tx_var = NULL;
+	data->maxdigits = 0;
+
+	if (!strcasecmp(args.direction, "tx")) {
+		data->tx_var = ast_strdup(args.varname);
+	} else if (!strcasecmp(args.direction, "rx")) {
+		data->rx_var = ast_strdup(args.varname);
+	} else {
+		ast_log(LOG_ERROR, "Direction must be either RX or TX\n");
+		return -1;
+	}
+
+	if (!ast_strlen_zero(args.maxdigits)) {
+		if (ast_str_to_int(args.maxdigits,&(data->maxdigits))) {
+			ast_log(LOG_ERROR, "Invalid integer: %s\n", args.maxdigits);
+			return -1;
+		}
+		if (data->maxdigits < 0) {
+			ast_log(LOG_ERROR, "Invalid natural number: %d\n", data->maxdigits);
+			return -1;
+		} else if (data->maxdigits == 0) {
+			ast_log(LOG_WARNING, "No maximum digit count set\n");
+		}
+	}
+
+	data->framehook_id = ast_framehook_attach(chan, &digit_framehook_interface);
+	if (data->framehook_id < 0) {
+		ast_log(AST_LOG_WARNING, "Failed to attach StoreDTMF framehook to '%s'\n",
+		        ast_channel_name(chan));
+		ast_datastore_free(datastore);
+		ast_free(data);
+		return -1;
+	}
+	datastore->data = data;
+
+	ast_channel_datastore_add(chan, datastore);
+
+	return 0;
+}
+
+static int unload_module(void)
+{
+	return ast_unregister_application(app);
+}
+
+static int load_module(void)
+{
+	return ast_register_application_xml(app, dtmfstore_exec);
+}
+
+AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "Technology independent async DTMF storage");

apps/app_externalivr.c

@@ -254,9 +254,9 @@ static struct ast_frame *gen_readframe(struct gen_state *state)
 	if (!(state->stream && (f = ast_readframe(state->stream)))) {
 		if (state->current) {
 			/* remove finished file from playlist */
-                        AST_LIST_LOCK(&u->playlist);
-                        AST_LIST_REMOVE_HEAD(&u->playlist, list);
-                        AST_LIST_UNLOCK(&u->playlist);
+			AST_LIST_LOCK(&u->playlist);
+			AST_LIST_REMOVE_HEAD(&u->playlist, list);
+			AST_LIST_UNLOCK(&u->playlist);
 			/* add finished file to finishlist */
 			AST_LIST_LOCK(&u->finishlist);
 			AST_LIST_INSERT_TAIL(&u->finishlist, state->current, list);
@@ -581,7 +581,7 @@ static int app_exec(struct ast_channel *chan, const char *data)
 		}
 	}
 
-	exit:
+  exit:
 	if (u->gen_active) {
 		ast_deactivate_generator(chan);
 	}
@@ -622,21 +622,21 @@ static int app_exec(struct ast_channel *chan, const char *data)
 }
 
 static int eivr_comm(struct ast_channel *chan, struct ivr_localuser *u,
-				struct ast_iostream *eivr_events,
-				struct ast_iostream *eivr_commands,
-				struct ast_iostream *eivr_errors,
-				const struct ast_str *args, const struct ast_flags flags)
+	struct ast_iostream *eivr_events,
+	struct ast_iostream *eivr_commands,
+	struct ast_iostream *eivr_errors,
+	const struct ast_str *args, const struct ast_flags flags)
 {
 	char input[1024];
 	struct playlist_entry *entry;
 	struct ast_frame *f;
 	int ms;
- 	int exception;
- 	int ready_fd;
+	int exception;
+	int ready_fd;
 	int waitfds[2];
 	int r;
- 	struct ast_channel *rchan;
- 	int res = -1;
+	struct ast_channel *rchan;
+	int res = -1;
 	int hangup_info_sent = 0;
 
 	waitfds[0] = ast_iostream_get_fd(eivr_commands);
@@ -645,78 +645,78 @@ static int eivr_comm(struct ast_channel *chan, struct ivr_localuser *u,
 	while (1) {
 		if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_ZOMBIE)) {
 			ast_chan_log(LOG_ERROR, chan, "Is a zombie\n");
- 			break;
- 		}
- 		if (!hangup_info_sent && !(ast_test_flag(&flags, run_dead)) && ast_check_hangup(chan)) {
+			break;
+		}
+		if (!hangup_info_sent && !(ast_test_flag(&flags, run_dead)) && ast_check_hangup(chan)) {
 			if (ast_test_flag(&flags, ignore_hangup)) {
 				ast_verb(3, "Got check_hangup, but ignore_hangup set so sending 'I' command\n");
 				send_eivr_event(eivr_events, 'I', "HANGUP", chan);
 				hangup_info_sent = 1;
 			} else {
- 				ast_verb(3, "Got check_hangup\n");
- 				send_eivr_event(eivr_events, 'H', NULL, chan);
-	 			break;
+				ast_verb(3, "Got check_hangup\n");
+				send_eivr_event(eivr_events, 'H', NULL, chan);
+				break;
 			}
- 		}
+		}
 
- 		ready_fd = 0;
- 		ms = 100;
- 		errno = 0;
- 		exception = 0;
+		ready_fd = 0;
+		ms = 100;
+		errno = 0;
+		exception = 0;
 
 		rchan = ast_waitfor_nandfds(&chan, 1, waitfds, (eivr_errors) ? 2 : 1, &exception, &ready_fd, &ms);
 
- 		if (ast_channel_state(chan) == AST_STATE_UP && !AST_LIST_EMPTY(&u->finishlist)) {
- 			AST_LIST_LOCK(&u->finishlist);
- 			while ((entry = AST_LIST_REMOVE_HEAD(&u->finishlist, list))) {
- 				send_eivr_event(eivr_events, 'F', entry->filename, chan);
- 				ast_free(entry);
- 			}
- 			AST_LIST_UNLOCK(&u->finishlist);
- 		}
+		if (ast_channel_state(chan) == AST_STATE_UP && !AST_LIST_EMPTY(&u->finishlist)) {
+			AST_LIST_LOCK(&u->finishlist);
+			while ((entry = AST_LIST_REMOVE_HEAD(&u->finishlist, list))) {
+				send_eivr_event(eivr_events, 'F', entry->filename, chan);
+				ast_free(entry);
+			}
+			AST_LIST_UNLOCK(&u->finishlist);
+		}
 
- 		if (ast_channel_state(chan) == AST_STATE_UP && !(ast_check_hangup(chan)) && rchan) {
- 			/* the channel has something */
- 			f = ast_read(chan);
- 			if (!f) {
- 				ast_verb(3, "Returned no frame\n");
- 				send_eivr_event(eivr_events, 'H', NULL, chan);
- 				break;
- 			}
- 			if (f->frametype == AST_FRAME_DTMF) {
- 				send_eivr_event(eivr_events, f->subclass.integer, NULL, chan);
- 				if (u->option_autoclear) {
-  					AST_LIST_LOCK(&u->playlist);
-  					if (!u->abort_current_sound && !u->playing_silence) {
+		if (ast_channel_state(chan) == AST_STATE_UP && !(ast_check_hangup(chan)) && rchan) {
+			/* the channel has something */
+			f = ast_read(chan);
+			if (!f) {
+				ast_verb(3, "Returned no frame\n");
+				send_eivr_event(eivr_events, 'H', NULL, chan);
+				break;
+			}
+			if (f->frametype == AST_FRAME_DTMF) {
+				send_eivr_event(eivr_events, f->subclass.integer, NULL, chan);
+				if (u->option_autoclear) {
+					AST_LIST_LOCK(&u->playlist);
+					if (!u->abort_current_sound && !u->playing_silence) {
 						/* send interrupted file as T data */
- 						if ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
-	 						send_eivr_event(eivr_events, 'T', entry->filename, chan);
+						if ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
+							send_eivr_event(eivr_events, 'T', entry->filename, chan);
 							ast_free(entry);
 						}
 					}
-  					while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
- 						send_eivr_event(eivr_events, 'D', entry->filename, chan);
-  						ast_free(entry);
-  					}
-  					if (!u->playing_silence)
-  						u->abort_current_sound = 1;
-  					AST_LIST_UNLOCK(&u->playlist);
-  				}
- 			} else if ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP)) {
- 				ast_verb(3, "Got AST_CONTROL_HANGUP\n");
- 				send_eivr_event(eivr_events, 'H', NULL, chan);
+					while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
+						send_eivr_event(eivr_events, 'D', entry->filename, chan);
+						ast_free(entry);
+					}
+					if (!u->playing_silence)
+						u->abort_current_sound = 1;
+					AST_LIST_UNLOCK(&u->playlist);
+				}
+			} else if ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP)) {
+				ast_verb(3, "Got AST_CONTROL_HANGUP\n");
+				send_eivr_event(eivr_events, 'H', NULL, chan);
 				if (f->data.uint32) {
 					ast_channel_hangupcause_set(chan, f->data.uint32);
 				}
- 				ast_frfree(f);
- 				break;
- 			}
- 			ast_frfree(f);
- 		} else if (ready_fd == waitfds[0]) {
- 			if (exception) {
- 				ast_chan_log(LOG_ERROR, chan, "Child process went away\n");
-  				break;
-  			}
+				ast_frfree(f);
+				break;
+			}
+			ast_frfree(f);
+		} else if (ready_fd == waitfds[0]) {
+			if (exception) {
+				ast_chan_log(LOG_ERROR, chan, "Child process went away\n");
+				break;
+			}
 
 			r = ast_iostream_gets(eivr_commands, input, sizeof(input));
 			if (r <= 0) {
@@ -784,112 +784,112 @@ static int eivr_comm(struct ast_channel *chan, struct ivr_localuser *u,
 					u->abort_current_sound = 1;
 				}
 				AST_LIST_UNLOCK(&u->playlist);
- 			} else if (input[0] == EIVR_CMD_SQUE) {
+			} else if (input[0] == EIVR_CMD_SQUE) {
 				if (ast_channel_state(chan) != AST_STATE_UP || ast_check_hangup(chan)) {
 					ast_chan_log(LOG_WARNING, chan, "Queue re'S'et called on unanswered channel\n");
 					send_eivr_event(eivr_events, 'Z', NULL, chan);
 					continue;
 				}
- 				if (!ast_fileexists(&input[2], NULL, ast_channel_language(u->chan))) {
- 					ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
- 					send_eivr_event(eivr_events, 'Z', &input[2], chan);
- 				} else {
- 					AST_LIST_LOCK(&u->playlist);
-	 				if (!u->abort_current_sound && !u->playing_silence) {
+				if (!ast_fileexists(&input[2], NULL, ast_channel_language(u->chan))) {
+					ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
+					send_eivr_event(eivr_events, 'Z', &input[2], chan);
+				} else {
+					AST_LIST_LOCK(&u->playlist);
+					if (!u->abort_current_sound && !u->playing_silence) {
 						/* send interrupted file as T data */
- 						if ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
-	 						send_eivr_event(eivr_events, 'T', entry->filename, chan);
+						if ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
+							send_eivr_event(eivr_events, 'T', entry->filename, chan);
 							ast_free(entry);
 						}
 					}
- 					while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
- 						send_eivr_event(eivr_events, 'D', entry->filename, chan);
- 						ast_free(entry);
+					while ((entry = AST_LIST_REMOVE_HEAD(&u->playlist, list))) {
+						send_eivr_event(eivr_events, 'D', entry->filename, chan);
+						ast_free(entry);
 					}
-	 				if (!u->playing_silence) {
- 						u->abort_current_sound = 1;
+					if (!u->playing_silence) {
+						u->abort_current_sound = 1;
 					}
- 					entry = make_entry(&input[2]);
- 					if (entry) {
- 						AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
+					entry = make_entry(&input[2]);
+					if (entry) {
+						AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
 					}
-	 				AST_LIST_UNLOCK(&u->playlist);
+					AST_LIST_UNLOCK(&u->playlist);
 				}
- 			} else if (input[0] == EIVR_CMD_APND) {
+			} else if (input[0] == EIVR_CMD_APND) {
 				if (ast_channel_state(chan) != AST_STATE_UP || ast_check_hangup(chan)) {
 					ast_chan_log(LOG_WARNING, chan, "Queue 'A'ppend called on unanswered channel\n");
 					send_eivr_event(eivr_events, 'Z', NULL, chan);
 					continue;
 				}
- 				if (!ast_fileexists(&input[2], NULL, ast_channel_language(u->chan))) {
- 					ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
- 					send_eivr_event(eivr_events, 'Z', &input[2], chan);
- 				} else {
-	 				entry = make_entry(&input[2]);
- 					if (entry) {
- 						AST_LIST_LOCK(&u->playlist);
- 						AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
- 						AST_LIST_UNLOCK(&u->playlist);
+				if (!ast_fileexists(&input[2], NULL, ast_channel_language(u->chan))) {
+					ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
+					send_eivr_event(eivr_events, 'Z', &input[2], chan);
+				} else {
+					entry = make_entry(&input[2]);
+					if (entry) {
+						AST_LIST_LOCK(&u->playlist);
+						AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
+						AST_LIST_UNLOCK(&u->playlist);
 					}
- 				}
- 			} else if (input[0] == EIVR_CMD_GET) {
- 				char response[2048];
+				}
+			} else if (input[0] == EIVR_CMD_GET) {
+				char response[2048];
 				ast_verb(4, "Retriving Variables from channel: %s\n", &input[2]);
- 				ast_eivr_getvariable(chan, &input[2], response, sizeof(response));
- 				send_eivr_event(eivr_events, 'G', response, chan);
- 			} else if (input[0] == EIVR_CMD_SVAR) {
+				ast_eivr_getvariable(chan, &input[2], response, sizeof(response));
+				send_eivr_event(eivr_events, 'G', response, chan);
+			} else if (input[0] == EIVR_CMD_SVAR) {
 				ast_verb(4, "Setting Variables in channel: %s\n", &input[2]);
- 				ast_eivr_setvariable(chan, &input[2]);
- 			} else if (input[0] == EIVR_CMD_LOG) {
- 				ast_chan_log(LOG_NOTICE, chan, "Log message from EIVR: %s\n", &input[2]);
- 			} else if (input[0] == EIVR_CMD_XIT) {
- 				ast_chan_log(LOG_NOTICE, chan, "Exiting: %s\n", &input[2]);
+				ast_eivr_setvariable(chan, &input[2]);
+			} else if (input[0] == EIVR_CMD_LOG) {
+				ast_chan_log(LOG_NOTICE, chan, "Log message from EIVR: %s\n", &input[2]);
+			} else if (input[0] == EIVR_CMD_XIT) {
+				ast_chan_log(LOG_NOTICE, chan, "Exiting: %s\n", &input[2]);
 				ast_chan_log(LOG_WARNING, chan, "e'X'it command is depricated, use 'E'xit instead\n");
- 				res = 0;
- 				break;
+				res = 0;
+				break;
 			} else if (input[0] == EIVR_CMD_EXIT) {
- 				ast_chan_log(LOG_NOTICE, chan, "Exiting: %s\n", &input[2]);
- 				send_eivr_event(eivr_events, 'E', NULL, chan);
- 				res = 0;
- 				break;
- 			} else if (input[0] == EIVR_CMD_HGUP) {
- 				ast_chan_log(LOG_NOTICE, chan, "Hanging up: %s\n", &input[2]);
- 				send_eivr_event(eivr_events, 'H', NULL, chan);
- 				break;
- 			} else if (input[0] == EIVR_CMD_OPT) {
+				ast_chan_log(LOG_NOTICE, chan, "Exiting: %s\n", &input[2]);
+				send_eivr_event(eivr_events, 'E', NULL, chan);
+				res = 0;
+				break;
+			} else if (input[0] == EIVR_CMD_HGUP) {
+				ast_chan_log(LOG_NOTICE, chan, "Hanging up: %s\n", &input[2]);
+				send_eivr_event(eivr_events, 'H', NULL, chan);
+				break;
+			} else if (input[0] == EIVR_CMD_OPT) {
 				if (ast_channel_state(chan) != AST_STATE_UP || ast_check_hangup(chan)) {
 					ast_chan_log(LOG_WARNING, chan, "Option called on unanswered channel\n");
 					send_eivr_event(eivr_events, 'Z', NULL, chan);
 					continue;
 				}
- 				if (!strcasecmp(&input[2], "autoclear"))
- 					u->option_autoclear = 1;
- 				else if (!strcasecmp(&input[2], "noautoclear"))
- 					u->option_autoclear = 0;
- 				else
- 					ast_chan_log(LOG_WARNING, chan, "Unknown option requested: %s\n", &input[2]);
- 			}
- 		} else if (ready_fd == waitfds[1]) {
- 			if (exception) {
- 				ast_chan_log(LOG_ERROR, chan, "Child process went away\n");
- 				break;
- 			}
-
- 			r = ast_iostream_gets(eivr_errors, input, sizeof(input));
- 			if (r > 0) {
- 				ast_chan_log(LOG_NOTICE, chan, "stderr: %s\n", ast_strip(input));
- 			} else if (r == 0) {
- 				ast_chan_log(LOG_ERROR, chan, "Child process went away\n");
- 				break;
+				if (!strcasecmp(&input[2], "autoclear"))
+					u->option_autoclear = 1;
+				else if (!strcasecmp(&input[2], "noautoclear"))
+					u->option_autoclear = 0;
+				else
+					ast_chan_log(LOG_WARNING, chan, "Unknown option requested: %s\n", &input[2]);
+			}
+		} else if (ready_fd == waitfds[1]) {
+			if (exception) {
+				ast_chan_log(LOG_ERROR, chan, "Child process went away\n");
+				break;
 			}
- 		} else if ((ready_fd < 0) && ms) {
- 			if (errno == 0 || errno == EINTR)
- 				continue;
 
- 			ast_chan_log(LOG_ERROR, chan, "Wait failed (%s)\n", strerror(errno));
- 			break;
- 		}
- 	}
+			r = ast_iostream_gets(eivr_errors, input, sizeof(input));
+			if (r > 0) {
+				ast_chan_log(LOG_NOTICE, chan, "stderr: %s\n", ast_strip(input));
+			} else if (r == 0) {
+				ast_chan_log(LOG_ERROR, chan, "Child process went away\n");
+				break;
+			}
+		} else if ((ready_fd < 0) && ms) {
+			if (errno == 0 || errno == EINTR)
+				continue;
+
+			ast_chan_log(LOG_ERROR, chan, "Wait failed (%s)\n", strerror(errno));
+			break;
+		}
+	}
 
 	return res;
 }

apps/app_fax.c

@@ -13,13 +13,17 @@
  */
 
 /*** MODULEINFO
-	<defaultenabled>no</defaultenabled>
 	<depend>spandsp</depend>
 	<conflict>res_fax</conflict>
+	<defaultenabled>no</defaultenabled>
 	<support_level>deprecated</support_level>
 	<replacement>res_fax</replacement>
+	<deprecated_in>16</deprecated_in>
+	<removed_in>19</removed_in>
 ***/
 
+/* Needed for spandsp headers */
+#define ASTMM_LIBC ASTMM_IGNORE
 #include "asterisk.h"
 
 #include <string.h>

apps/app_festival.c

@@ -281,8 +281,6 @@ static int festival_exec(struct ast_channel *chan, const char *vdata)
 	int usecache;
 	int res = 0;
 	struct sockaddr_in serv_addr;
-	struct hostent *serverhost;
-	struct ast_hostent ahp;
 	int fd;
 	FILE *fs;
 	const char *host;
@@ -398,15 +396,17 @@ static int festival_exec(struct ast_channel *chan, const char *vdata)
 
 	if ((serv_addr.sin_addr.s_addr = inet_addr(host)) == -1) {
 		/* its a name rather than an ipnum */
-		serverhost = ast_gethostbyname(host, &ahp);
+		struct ast_sockaddr addr = { {0,} };
 
-		if (serverhost == NULL) {
-			ast_log(LOG_WARNING, "festival_client: gethostbyname failed\n");
+		if (ast_sockaddr_resolve_first_af(&addr, host, PARSE_PORT_FORBID, AF_INET)) {
+			ast_log(LOG_WARNING, "festival_client: ast_sockaddr_resolve_first_af() failed\n");
 			ast_config_destroy(cfg);
 			close(fd);
 			return -1;
 		}
-		memmove(&serv_addr.sin_addr, serverhost->h_addr, serverhost->h_length);
+
+		/* We'll overwrite port and family in a sec */
+		ast_sockaddr_to_sin(&addr, &serv_addr);
 	}
 
 	serv_addr.sin_family = AF_INET;
@@ -433,7 +433,7 @@ static int festival_exec(struct ast_channel *chan, const char *vdata)
 	}
 	readcache = 0;
 	writecache = 0;
-	if (strlen(cachedir) + strlen(MD5Hex) + 1 <= MAXFESTLEN && (usecache == -1)) {
+	if (strlen(cachedir) + sizeof(MD5Hex) + 1 <= MAXFESTLEN && (usecache == -1)) {
 		snprintf(cachefile, sizeof(cachefile), "%s/%s", cachedir, MD5Hex);
 		fdesc = open(cachefile, O_RDWR);
 		if (fdesc == -1) {

apps/app_followme.c

@@ -224,7 +224,7 @@ struct findme_user {
 	long digts;
 	int ynidx;
 	int state;
-	char dialarg[256];
+	char dialarg[768];
 	/*! Collected digits to accept/decline the call. */
 	char yn[MAX_YN_STRING];
 	/*! TRUE if the outgoing call is answered. */
@@ -408,7 +408,6 @@ static int reload_followme(int reload)
 	char *cat = NULL, *tmp;
 	struct ast_variable *var;
 	struct number *cur, *nm;
-	char *numberstr;
 	int timeout;
 	int numorder;
 	const char* enable_callee_prompt_str;
@@ -536,9 +535,12 @@ static int reload_followme(int reload)
 		while (var) {
 			if (!strcasecmp(var->name, "number")) {
 				int idx = 0;
+				char copy[strlen(var->value) + 1];
+				char *numberstr;
 
 				/* Add a new number */
-				numberstr = ast_strdupa(var->value);
+				strcpy(copy, var->value); /* safe */
+				numberstr = copy;
 				if ((tmp = strchr(numberstr, ','))) {
 					*tmp++ = '\0';
 					timeout = atoi(tmp);
@@ -762,10 +764,6 @@ static struct ast_channel *wait_for_winner(struct findme_user_listptr *findme_us
 		}
 
 		tmpto = to;
-		if (to < 0) {
-			to = 1000;
-			tmpto = 1000;
-		}
 		towas = to;
 		winner = ast_waitfor_n(watchers, pos, &to);
 		tmpto -= to;
@@ -915,6 +913,11 @@ static struct ast_channel *wait_for_winner(struct findme_user_listptr *findme_us
 						/* Ignore going off hook and flash */
 						break;
 					case AST_CONTROL_CONNECTED_LINE:
+						if (ast_test_flag(&tpargs->followmeflags, FOLLOWMEFLAG_IGNORE_CONNECTEDLINE)) {
+							ast_verb(3, "Connected line update from %s prevented.\n",
+								ast_channel_name(winner));
+							break;
+						}
 						if (!tmpuser) {
 							/*
 							 * Hold connected line update from caller until we have a
@@ -930,11 +933,6 @@ static struct ast_channel *wait_for_winner(struct findme_user_listptr *findme_us
 								tpargs->pending_in_connected_update = 1;
 							}
 							ast_party_connected_line_free(&connected);
-							break;
-						}
-						if (ast_test_flag(&tpargs->followmeflags, FOLLOWMEFLAG_IGNORE_CONNECTEDLINE)) {
-							ast_verb(3, "Connected line update from %s prevented.\n",
-								ast_channel_name(winner));
 						} else {
 							ast_verb(3,
 								"%s connected line has changed. Saving it until answer.\n",

apps/app_forkcdr.c

@@ -2,7 +2,7 @@
  * Asterisk -- An open source telephony toolkit.
  *
  * Copyright (C) 1999 - 2005, Anthony Minessale anthmct@yahoo.com
- * Development of this app Sponsered/Funded  by TAAN Softworks Corp
+ * Development of this app Sponsored/Funded  by TAAN Softworks Corp
  *
  * See http://www.asterisk.org for more information about
  * the Asterisk project. Please do not directly contact

apps/app_getcpeid.c

@@ -26,6 +26,7 @@
  */
 
 /*** MODULEINFO
+	<depend>res_adsi</depend>
 	<support_level>deprecated</support_level>
  ***/
 

apps/app_ices.c

@@ -29,6 +29,8 @@
 
 /*** MODULEINFO
 	<support_level>deprecated</support_level>
+	<deprecated_in>16</deprecated_in>
+	<removed_in>19</removed_in>
  ***/
 
 #include "asterisk.h"

apps/app_image.c

@@ -27,6 +27,8 @@
 
 /*** MODULEINFO
 	<support_level>deprecated</support_level>
+	<deprecated_in>16</deprecated_in>
+	<removed_in>19</removed_in>
  ***/
 
 #include "asterisk.h"

apps/app_jack.c

@@ -615,8 +615,6 @@ static int queue_voice_frame(struct jack_data *jack_data, struct ast_frame *f)
  * Read data from the input ringbuffer, which is the properly resampled audio
  * that was read from the jack input port.  Write it to the channel in 20 ms frames,
  * or fill up an output frame instead if one is provided.
- *
- * \return Nothing.
  */
 static void handle_jack_audio(struct ast_channel *chan, struct jack_data *jack_data,
 	struct ast_frame *out_frame)

apps/app_macro.c

@@ -29,6 +29,8 @@
 	<defaultenabled>no</defaultenabled>
 	<support_level>deprecated</support_level>
 	<replacement>app_stack (GoSub)</replacement>
+	<deprecated_in>16</deprecated_in>
+	<removed_in>21</removed_in>
  ***/
 
 #include "asterisk.h"
@@ -73,7 +75,7 @@
 			of nesting (macro calling macro calling macro, etc.); It may be possible that stack-intensive
 			applications in deeply nested macros could cause asterisk to crash earlier than this limit.
 			It is advised that if you need to deeply nest macro calls, that you use the Gosub application
-			(now allows arguments like a Macro) with explict Return() calls instead.</para></warning>
+			(now allows arguments like a Macro) with explicit Return() calls instead.</para></warning>
 			<warning><para>Use of the application <literal>WaitExten</literal> within a macro will not function
 			as expected. Please use the <literal>Read</literal> application in order to read DTMF from a channel
 			currently executing a macro.</para></warning>

apps/app_meetme.c

@@ -43,6 +43,8 @@
 	<defaultenabled>no</defaultenabled>
 	<support_level>extended</support_level>
 	<replacement>app_confbridge</replacement>
+	<deprecated_in>19</deprecated_in>
+	<removed_in>21</removed_in>
  ***/
 
 #include "asterisk.h"
@@ -637,6 +639,82 @@
 			</syntax>
 		</managerEventInstance>
 	</managerEvent>
+	<managerEvent language="en_US" name="MeetmeList">
+		<managerEventInstance class="EVENT_FLAG_CALL">
+			<synopsis>Raised in response to a MeetmeList command.</synopsis>
+			<syntax>
+				<parameter name="Conference">
+					<para>Conference ID.</para>
+				</parameter>
+				<parameter name="UserNumber">
+					<para>User ID.</para>
+				</parameter>
+				<parameter name="CallerIDNum">
+					<para>Caller ID number.</para>
+				</parameter>
+				<parameter name="CallerIDName">
+					<para>Caller ID name.</para>
+				</parameter>
+				<parameter name="ConnectedLineNum">
+					<para>Connected Line number.</para>
+				</parameter>
+				<parameter name="ConnectedLineName">
+					<para>Connected Line name.</para>
+				</parameter>
+				<parameter name="Channel">
+					<para>Channel name</para>
+				</parameter>
+				<parameter name="Admin">
+					<para>Whether or not the user is an admin.</para>
+				</parameter>
+				<parameter name="Role">
+					<para>User role. Can be "Listen only", "Talk only", or "Talk and listen".</para>
+				</parameter>
+				<parameter name="MarkedUser">
+					<para>Whether or not the user is a marked user.</para>
+				</parameter>
+				<parameter name="Muted">
+					<para>Whether or not the user is currently muted.</para>
+				</parameter>
+				<parameter name="Talking">
+					<para>Whether or not the user is currently talking.</para>
+				</parameter>
+			</syntax>
+			<see-also>
+				<ref type="manager">MeetmeList</ref>
+				<ref type="application">MeetMe</ref>
+			</see-also>
+		</managerEventInstance>
+	</managerEvent>
+	<managerEvent language="en_US" name="MeetmeListRooms">
+		<managerEventInstance class="EVENT_FLAG_CALL">
+			<synopsis>Raised in response to a MeetmeListRooms command.</synopsis>
+			<syntax>
+				<parameter name="Conference">
+					<para>Conference ID.</para>
+				</parameter>
+				<parameter name="Parties">
+					<para>Number of parties in the conference.</para>
+				</parameter>
+				<parameter name="Marked">
+					<para>Number of marked users in the conference.</para>
+				</parameter>
+				<parameter name="Activity">
+					<para>Total duration of conference in HH:MM:SS format.</para>
+				</parameter>
+				<parameter name="Creation">
+					<para>How the conference was created: "Dyanmic" or "Static".</para>
+				</parameter>
+				<parameter name="Locked">
+					<para>Whether or not the conference is locked.</para>
+				</parameter>
+			</syntax>
+			<see-also>
+				<ref type="manager">MeetmeListRooms</ref>
+				<ref type="application">MeetMe</ref>
+			</see-also>
+		</managerEventInstance>
+	</managerEvent>
  ***/
 
 #define CONFIG_FILE_NAME	"meetme.conf"
@@ -1334,7 +1412,7 @@ static struct ast_json *status_to_json(int on)
  * \brief Generate a stasis message associated with a meetme event
  * \since 12.0.0
  *
- * \param meetme_confere The conference responsible for generating this message
+ * \param meetme_conference The conference responsible for generating this message
  * \param chan The channel involved in the message (NULL allowed)
  * \param user The conference user involved in the message (NULL allowed)
  * \param message_type the type the stasis message being generated
@@ -1624,8 +1702,14 @@ static struct ast_conference *build_conf(const char *confno, const char *pin,
 
 	ast_format_cap_append(cap_slin, ast_format_slin, 0);
 	/* Make a new one */
-	if (!(cnf = ast_calloc(1, sizeof(*cnf))) ||
-		!(cnf->usercontainer = ao2_container_alloc(1, NULL, user_no_cmp))) {
+	cnf = ast_calloc(1, sizeof(*cnf));
+	if (!cnf) {
+		goto cnfout;
+	}
+
+	cnf->usercontainer = ao2_container_alloc_list(AO2_ALLOC_OPT_LOCK_MUTEX, 0,
+		NULL, user_no_cmp);
+	if (!cnf->usercontainer) {
 		goto cnfout;
 	}
 
@@ -2828,7 +2912,7 @@ static void meetme_menu_normal(enum menu_modes *menu_mode, int *dtmf, struct ast
 }
 
 /*! \internal
- * \brief Processes menu options for the adminstrator menu (accessible through the 's' option for app_meetme)
+ * \brief Processes menu options for the administrator menu (accessible through the 's' option for app_meetme)
  *
  * \param menu_mode a pointer to the currently active menu_mode.
  * \param dtmf a pointer to the dtmf value currently being processed against the menu.
@@ -2955,7 +3039,8 @@ static void meetme_menu_admin(enum menu_modes *menu_mode, int *dtmf, struct ast_
  * \param confflags flags used by conf for various options
  * \param chan ast_channel belonging to the user who called the menu
  * \param user which meetme conference user invoked the menu
- * \param recordingtmp character buffer which may hold the name of the conference recording file
+ * \param recordingtmp, recordingtmp_size character buffer which may hold the name of the conference recording file
+ * \param cap_slin
  */
 static void meetme_menu_admin_extended(enum menu_modes *menu_mode, int *dtmf,
 	struct ast_conference *conf, struct ast_flags64 *confflags, struct ast_channel *chan,
@@ -3142,7 +3227,8 @@ static void meetme_menu_admin_extended(enum menu_modes *menu_mode, int *dtmf,
  * \param confflags flags used by conf for various options
  * \param chan ast_channel belonging to the user who called the menu
  * \param user which meetme conference user invoked the menu
- * \param recordingtmp character buffer which may hold the name of the conference recording file
+ * \param recordingtmp,recordingtmp_size character buffer which may hold the name of the conference recording file
+ * \param cap_slin
  */
 static void meetme_menu(enum menu_modes *menu_mode, int *dtmf,
 	struct ast_conference *conf, struct ast_flags64 *confflags, struct ast_channel *chan,
@@ -3424,7 +3510,9 @@ static int conf_run(struct ast_channel *chan, struct ast_conference *conf, struc
 	user->chan = chan;
 	user->userflags = *confflags;
 	user->adminflags = ast_test_flag64(confflags, CONFFLAG_STARTMUTED) ? ADMINFLAG_SELFMUTED : 0;
-	user->adminflags |= (conf->gmuted) ? ADMINFLAG_MUTED : 0;
+	if (!ast_test_flag64(confflags, CONFFLAG_ADMIN)) {
+		user->adminflags |= (conf->gmuted) ? ADMINFLAG_MUTED : 0;
+	}
 	user->talking = -1;
 
 	ast_mutex_unlock(&conf->playlock);
@@ -4749,7 +4837,7 @@ static struct ast_conference *find_conf(struct ast_channel *chan, char *confno,
 				}
 			}
 			if (!var) {
-				ast_debug(1, "%s isn't a valid conference\n", confno);
+				ast_log(LOG_WARNING, "%s isn't a valid conference\n", confno);
 			}
 			ast_config_destroy(cfg);
 		}
@@ -5211,7 +5299,7 @@ static int user_chan_cb(void *obj, void *args, int flags)
 	return 0;
 }
 
-/*! \brief The MeetMeadmin application
+/*! \brief The MeetMeAdmin application
 
   MeetMeAdmin(confno, command, caller) */
 static int admin_exec(struct ast_channel *chan, const char *data) {
@@ -5227,7 +5315,9 @@ static int admin_exec(struct ast_channel *chan, const char *data) {
 
 	if (ast_strlen_zero(data)) {
 		ast_log(LOG_WARNING, "MeetMeAdmin requires an argument!\n");
-		pbx_builtin_setvar_helper(chan, "MEETMEADMINSTATUS", "NOPARSE");
+		if (chan) {
+			pbx_builtin_setvar_helper(chan, "MEETMEADMINSTATUS", "NOPARSE");
+		}
 		return -1;
 	}
 
@@ -5236,7 +5326,9 @@ static int admin_exec(struct ast_channel *chan, const char *data) {
 
 	if (!args.command) {
 		ast_log(LOG_WARNING, "MeetmeAdmin requires a command!\n");
-		pbx_builtin_setvar_helper(chan, "MEETMEADMINSTATUS", "NOPARSE");
+		if (chan) {
+			pbx_builtin_setvar_helper(chan, "MEETMEADMINSTATUS", "NOPARSE");
+		}
 		return -1;
 	}
 
@@ -5249,7 +5341,9 @@ static int admin_exec(struct ast_channel *chan, const char *data) {
 	if (!cnf) {
 		ast_log(LOG_WARNING, "Conference number '%s' not found!\n", args.confno);
 		AST_LIST_UNLOCK(&confs);
-		pbx_builtin_setvar_helper(chan, "MEETMEADMINSTATUS", "NOTFOUND");
+		if (chan) {
+			pbx_builtin_setvar_helper(chan, "MEETMEADMINSTATUS", "NOTFOUND");
+		}
 		return 0;
 	}
 
@@ -5372,7 +5466,9 @@ usernotfound:
 	AST_LIST_UNLOCK(&confs);
 
 	dispose_conf(cnf);
-	pbx_builtin_setvar_helper(chan, "MEETMEADMINSTATUS", res == -2 ? "NOTFOUND" : res ? "FAILED" : "OK");
+	if (chan) {
+		pbx_builtin_setvar_helper(chan, "MEETMEADMINSTATUS", res == -2 ? "NOTFOUND" : res ? "FAILED" : "OK");
+	}
 
 	return 0;
 }
@@ -7395,13 +7491,6 @@ static void sla_station_destructor(void *obj)
 	ast_string_field_free_memory(station);
 }
 
-static int sla_trunk_hash(const void *obj, const int flags)
-{
-	const struct sla_trunk *trunk = obj;
-
-	return ast_str_case_hash(trunk->name);
-}
-
 static int sla_trunk_cmp(void *obj, void *arg, int flags)
 {
 	struct sla_trunk *trunk = obj, *trunk2 = arg;
@@ -7409,13 +7498,6 @@ static int sla_trunk_cmp(void *obj, void *arg, int flags)
 	return !strcasecmp(trunk->name, trunk2->name) ? CMP_MATCH | CMP_STOP : 0;
 }
 
-static int sla_station_hash(const void *obj, const int flags)
-{
-	const struct sla_station *station = obj;
-
-	return ast_str_case_hash(station->name);
-}
-
 static int sla_station_cmp(void *obj, void *arg, int flags)
 {
 	struct sla_station *station = obj, *station2 = arg;
@@ -7869,8 +7951,10 @@ static int sla_load_config(int reload)
 	if (!reload) {
 		ast_mutex_init(&sla.lock);
 		ast_cond_init(&sla.cond, NULL);
-		sla_trunks = ao2_container_alloc(1, sla_trunk_hash, sla_trunk_cmp);
-		sla_stations = ao2_container_alloc(1, sla_station_hash, sla_station_cmp);
+		sla_trunks = ao2_container_alloc_list(AO2_ALLOC_OPT_LOCK_MUTEX, 0,
+			NULL, sla_trunk_cmp);
+		sla_stations = ao2_container_alloc_list(AO2_ALLOC_OPT_LOCK_MUTEX, 0,
+			NULL, sla_station_cmp);
 	}
 
 	if (!(cfg = ast_config_load(SLA_CONFIG_FILE, config_flags))) {

apps/app_mf.c

@@ -0,0 +1,546 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2021, Naveen Albert
+ *
+ * Naveen Albert <asterisk@phreaknet.org>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief MF sender and receiver applications
+ *
+ * \author Naveen Albert <asterisk@phreaknet.org>
+ *
+ * \ingroup applications
+ */
+
+/*** MODULEINFO
+	<support_level>extended</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include "asterisk/file.h"
+#include "asterisk/pbx.h"
+#include "asterisk/channel.h"
+#include "asterisk/dsp.h"
+#include "asterisk/app.h"
+#include "asterisk/module.h"
+#include "asterisk/indications.h"
+#include "asterisk/conversions.h"
+
+/*** DOCUMENTATION
+	<application name="ReceiveMF" language="en_US">
+		<since>
+			<version>16.21.0</version>
+			<version>18.7.0</version>
+			<version>19.0.0</version>
+		</since>
+		<synopsis>
+			Detects MF digits on a channel and saves them to a variable.
+		</synopsis>
+		<syntax>
+			<parameter name="variable" required="true">
+				<para>The input digits will be stored in the given
+				<replaceable>variable</replaceable> name.</para>
+			</parameter>
+			<parameter name="timeout">
+				<para>The number of seconds to wait for all digits, if greater
+				than <literal>0</literal>. Can be floating point. Default
+				is no timeout.</para>
+			</parameter>
+			<parameter name="options">
+				<optionlist>
+					<option name="d">
+						<para>Delay audio by a frame to try to extra quelch.</para>
+					</option>
+					<option name="l">
+						<para>Receive digits even if a key pulse (KP) has not yet
+						been received. By default, this application will ignore
+						all other digits until a KP has been received.</para>
+					</option>
+					<option name="k">
+						<para>Do not return a character for the KP digit.</para>
+					</option>
+					<option name="m">
+						<para>Mute conference.</para>
+					</option>
+					<option name="n">
+						<para>Maximum number of digits, regardless of the sequence.</para>
+					</option>
+					<option name="o">
+						<para>Enable override. Repeated KPs will clear all previous digits.</para>
+					</option>
+					<option name="q">
+						<para>Quelch MF from in-band.</para>
+					</option>
+					<option name="r">
+						<para>"Radio" mode (relaxed MF).</para>
+					</option>
+					<option name="s">
+						<para>Do not return a character for ST digits.</para>
+					</option>
+				</optionlist>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Reads a ST, STP, ST2P, or ST3P-terminated string of MF digits from
+			the user in to the given <replaceable>variable</replaceable>.</para>
+			<para>This application does not automatically answer the channel and
+			should be preceded with <literal>Answer</literal> or
+			<literal>Progress</literal> as needed.</para>
+			<variablelist>
+				<variable name="RECEIVEMFSTATUS">
+					<para>This is the status of the read operation.</para>
+					<value name="START" />
+					<value name="ERROR" />
+					<value name="HANGUP" />
+					<value name="MAXDIGITS" />
+					<value name="TIMEOUT" />
+				</variable>
+			</variablelist>
+		</description>
+		<see-also>
+			<ref type="application">Read</ref>
+			<ref type="application">SendMF</ref>
+			<ref type="application">ReceiveSF</ref>
+		</see-also>
+	</application>
+	<application name="SendMF" language="en_US">
+		<since>
+			<version>16.21.0</version>
+			<version>18.7.0</version>
+			<version>19.0.0</version>
+		</since>
+		<synopsis>
+			Sends arbitrary MF digits on the current or specified channel.
+		</synopsis>
+		<syntax>
+			<parameter name="digits" required="true">
+				<para>List of digits 0-9,*#ABC to send; w for a half-second pause,
+				also f or F for a flash-hook if the channel supports flash-hook,
+				h or H for 250 ms of 2600 Hz,
+				and W for a wink if the channel supports wink.</para>
+				<para>Key pulse and start digits are not included automatically.
+				* is used for KP, # for ST, A for STP, B for ST2P, and C for ST3P.</para>
+			</parameter>
+			<parameter name="timeout_ms" required="false">
+				<para>Amount of time to wait in ms between tones. (defaults to 50ms).</para>
+			</parameter>
+			<parameter name="duration_ms" required="false">
+				<para>Duration of each numeric digit (defaults to 55ms).</para>
+			</parameter>
+			<parameter name="duration_ms_kp" required="false">
+				<para>Duration of KP digits (defaults to 120ms).</para>
+			</parameter>
+			<parameter name="duration_ms_st" required="false">
+				<para>Duration of ST, STP, ST2P, and ST3P digits (defaults to 65ms).</para>
+			</parameter>
+			<parameter name="channel" required="false">
+				<para>Channel where digits will be played</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>It will send all digits or terminate if it encounters an error.</para>
+		</description>
+		<see-also>
+			<ref type="application">ReceiveMF</ref>
+			<ref type="application">SendSF</ref>
+			<ref type="application">SendDTMF</ref>
+		</see-also>
+	</application>
+	<manager name="PlayMF" language="en_US">
+		<since>
+			<version>16.21.0</version>
+			<version>18.7.0</version>
+			<version>19.0.0</version>
+		</since>
+		<synopsis>
+			Play MF digit on a specific channel.
+		</synopsis>
+		<syntax>
+			<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
+			<parameter name="Channel" required="true">
+				<para>Channel name to send digit to.</para>
+			</parameter>
+			<parameter name="Digit" required="true">
+				<para>The MF digit to play.</para>
+			</parameter>
+			<parameter name="Duration" required="false">
+				<para>The duration, in milliseconds, of the digit to be played.</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Plays an MF digit on the specified channel.</para>
+		</description>
+	</manager>
+ ***/
+
+enum read_option_flags {
+	OPT_DELAY = (1 << 0),
+	OPT_MUTE = (1 << 1),
+	OPT_QUELCH = (1 << 2),
+	OPT_RELAXED = (1 << 3),
+	OPT_LAX_KP = (1 << 4),
+	OPT_PROCESS = (1 << 5),
+	OPT_NO_KP = (1 << 6),
+	OPT_NO_ST = (1 << 7),
+	OPT_KP_OVERRIDE = (1 << 8),
+	OPT_MAXDIGITS = (1 << 9),
+};
+
+enum {
+	OPT_ARG_MAXDIGITS,
+	/* Must be the last element */
+	OPT_ARG_ARRAY_SIZE,
+};
+
+AST_APP_OPTIONS(read_app_options, {
+	AST_APP_OPTION('d', OPT_DELAY),
+	AST_APP_OPTION('l', OPT_LAX_KP),
+	AST_APP_OPTION('k', OPT_NO_KP),
+	AST_APP_OPTION('m', OPT_MUTE),
+	AST_APP_OPTION_ARG('n', OPT_MAXDIGITS, OPT_ARG_MAXDIGITS),
+	AST_APP_OPTION('o', OPT_KP_OVERRIDE),
+	AST_APP_OPTION('p', OPT_PROCESS),
+	AST_APP_OPTION('q', OPT_QUELCH),
+	AST_APP_OPTION('r', OPT_RELAXED),
+	AST_APP_OPTION('s', OPT_NO_ST),
+});
+
+static const char *readmf_name = "ReceiveMF";
+static const char sendmf_name[] = "SendMF";
+
+#define MF_BETWEEN_MS 50
+#define MF_DURATION 55
+#define MF_KP_DURATION 120
+#define MF_ST_DURATION 65
+
+/*!
+ * \brief Detects MF digits on channel using DSP, terminated by ST, STP, ST2P, or ST3P
+ *
+ * \param chan channel on which to read digits
+ * \param buf Buffer in which to store digits
+ * \param buflen Size of buffer
+ * \param timeout ms to wait for all digits before giving up
+ * \param features Any additional DSP features to use
+ * \param override Start over if we receive additional KPs
+ * \param no_kp Don't include KP in the output
+ * \param no_st Don't include start digits in the output
+ * \param maxdigits If greater than 0, only read this many digits no matter what
+ *
+ * \retval 0 if successful
+ * \retval -1 if unsuccessful (including hangup).
+ */
+static int read_mf_digits(struct ast_channel *chan, char *buf, int buflen, int timeout, int features, int laxkp, int override, int no_kp, int no_st, int maxdigits) {
+	struct ast_dsp *dsp;
+	struct ast_frame *frame = NULL;
+	struct timeval start;
+	int remaining_time = timeout;
+	int digits_read = 0;
+	int is_start_digit = 0;
+	char *str = buf;
+	int res = 0;
+
+	if (!(dsp = ast_dsp_new())) {
+		ast_log(LOG_WARNING, "Unable to allocate DSP!\n");
+		pbx_builtin_setvar_helper(chan, "RECEIVEMFSTATUS", "ERROR");
+		return -1;
+	}
+	ast_dsp_set_features(dsp, DSP_FEATURE_DIGIT_DETECT);
+	ast_dsp_set_digitmode(dsp, DSP_DIGITMODE_MF | features);
+
+	start = ast_tvnow();
+	*str = 0; /* start with empty output buffer */
+
+	/* based on app_read and generic_fax_exec from res_fax */
+	while (timeout == 0 || remaining_time > 0) {
+		if (timeout > 0) {
+			remaining_time = ast_remaining_ms(start, timeout);
+			if (remaining_time <= 0) {
+				pbx_builtin_setvar_helper(chan, "RECEIVEMFSTATUS", "TIMEOUT");
+				break;
+			}
+		}
+		if ((maxdigits && digits_read >= maxdigits) || digits_read >= (buflen - 1)) { /* we don't have room to store any more digits (very unlikely to happen for a legitimate reason) */
+			/* This result will probably not be usable, so status should not be START */
+			pbx_builtin_setvar_helper(chan, "RECEIVEMFSTATUS", "MAXDIGITS");
+			break;
+		}
+		/* ast_waitfordigit only waits for DTMF frames, we need to do DSP on voice frames */
+		if (ast_waitfor(chan, 1000) > 0) {
+			frame = ast_read(chan);
+			if (!frame) {
+				ast_debug(1, "Channel '%s' did not return a frame; probably hung up.\n", ast_channel_name(chan));
+				pbx_builtin_setvar_helper(chan, "RECEIVEMFSTATUS", "HANGUP");
+				break;
+			} else if (frame->frametype == AST_FRAME_VOICE) {
+				frame = ast_dsp_process(chan, dsp, frame);
+				/* AST_FRAME_DTMF is used all over the DSP code for DTMF, MF, fax, etc.
+					It's used because we can use the frame to store the digit detected.
+					All this means is that we received something we care about. */
+				if (frame->frametype == AST_FRAME_DTMF) {
+					char result = frame->subclass.integer;
+					if (digits_read == 0 && !laxkp && result != '*') {
+						ast_debug(1, "Received MF digit, but no KP yet, ignoring: %c\n", result);
+						ast_frfree(frame);
+						continue;
+					}
+					ast_debug(1, "Received MF digit: %c\n", result);
+					if (result == '*') {
+						/* We received an additional KP, start over? */
+						if (override && digits_read > 0) {
+							ast_debug(1, "Received another KP, starting over\n");
+							str = buf;
+							*str = 0;
+							digits_read = 1; /* we just detected a KP */
+						} else {
+							digits_read++;
+						}
+						/* if we were told not to include the KP digit in the output string, then skip it */
+						if (no_kp) {
+							ast_frfree(frame);
+							continue;
+						}
+					} else {
+						digits_read++;
+					}
+					is_start_digit = (strchr("#", result) || strchr("A", result) || strchr("B", result) || strchr("C", result));
+					/* if we were told not to include the ST digit in the output string, then skip it */
+					if (!no_st || !is_start_digit) {
+						*str++ = result; /* won't write past allotted memory, because of buffer check at top of loop */
+						*str = 0;
+					}
+					/* we received a ST digit (ST, STP, ST2P, or ST3P), so we're done */
+					if (is_start_digit) {
+						pbx_builtin_setvar_helper(chan, "RECEIVEMFSTATUS", "START");
+						ast_frfree(frame);
+						break;
+					}
+					/* only free frame if it was a DSP match. The MF itself should not be muted. */
+					ast_frfree(frame);
+				}
+			}
+		} else {
+			pbx_builtin_setvar_helper(chan, "RECEIVEMFSTATUS", "HANGUP");
+			res = -1;
+		}
+	}
+	ast_dsp_free(dsp);
+	ast_debug(3, "channel '%s' - event loop stopped { timeout: %d, remaining_time: %d }\n", ast_channel_name(chan), timeout, remaining_time);
+	return res;
+}
+
+static int read_mf_exec(struct ast_channel *chan, const char *data)
+{
+#define BUFFER_SIZE 256
+	char tmp[BUFFER_SIZE] = "";
+	int to = 0;
+	double tosec;
+	struct ast_flags flags = {0};
+	char *optargs[OPT_ARG_ARRAY_SIZE];
+	char *argcopy = NULL;
+	int res, features = 0, maxdigits = 0;
+
+	AST_DECLARE_APP_ARGS(arglist,
+		AST_APP_ARG(variable);
+		AST_APP_ARG(timeout);
+		AST_APP_ARG(options);
+	);
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "ReceiveMF requires an argument (variable)\n");
+		return -1;
+	}
+
+	argcopy = ast_strdupa(data);
+
+	AST_STANDARD_APP_ARGS(arglist, argcopy);
+
+	if (!ast_strlen_zero(arglist.options)) {
+		ast_app_parse_options(read_app_options, &flags, optargs, arglist.options);
+	}
+
+	if (!ast_strlen_zero(arglist.timeout)) {
+		tosec = atof(arglist.timeout);
+		if (tosec <= 0) {
+			to = 0;
+		} else {
+			to = tosec * 1000.0;
+		}
+	}
+
+	if (ast_strlen_zero(arglist.variable)) {
+		ast_log(LOG_WARNING, "Invalid! Usage: ReceiveMF(variable[,timeout][,option])\n");
+		return -1;
+	}
+	if (ast_test_flag(&flags, OPT_MAXDIGITS) && !ast_strlen_zero(optargs[OPT_ARG_MAXDIGITS])) {
+		maxdigits = atoi(optargs[OPT_ARG_MAXDIGITS]);
+		if (maxdigits <= 0) {
+			ast_log(LOG_WARNING, "Invalid maximum number of digits, ignoring: '%s'\n", optargs[OPT_ARG_MAXDIGITS]);
+			maxdigits = 0;
+		}
+	}
+
+	if (ast_test_flag(&flags, OPT_DELAY)) {
+		features |= DSP_DIGITMODE_MUTEMAX;
+	}
+
+	if (ast_test_flag(&flags, OPT_MUTE)) {
+		features |= DSP_DIGITMODE_MUTECONF;
+	}
+
+	if (!ast_test_flag(&flags, OPT_QUELCH)) {
+		features |= DSP_DIGITMODE_NOQUELCH;
+	}
+
+	if (ast_test_flag(&flags, OPT_RELAXED)) {
+		features |= DSP_DIGITMODE_RELAXDTMF;
+	}
+
+	res = read_mf_digits(chan, tmp, BUFFER_SIZE, to, features, (ast_test_flag(&flags, OPT_LAX_KP)),
+		(ast_test_flag(&flags, OPT_KP_OVERRIDE)), (ast_test_flag(&flags, OPT_NO_KP)), (ast_test_flag(&flags, OPT_NO_ST)), maxdigits);
+	pbx_builtin_setvar_helper(chan, arglist.variable, tmp);
+	if (!ast_strlen_zero(tmp)) {
+		ast_verb(3, "MF digits received: '%s'\n", tmp);
+	} else if (!res) { /* if channel hung up, don't print anything out */
+		ast_verb(3, "No MF digits received.\n");
+	}
+	return res;
+}
+
+static int sendmf_exec(struct ast_channel *chan, const char *vdata)
+{
+	int res;
+	char *data;
+	int dinterval = 0, duration = 0, durationkp = 0, durationst = 0;
+	struct ast_channel *chan_found = NULL;
+	struct ast_channel *chan_dest = chan;
+	struct ast_channel *chan_autoservice = NULL;
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(digits);
+		AST_APP_ARG(dinterval);
+		AST_APP_ARG(duration);
+		AST_APP_ARG(durationkp);
+		AST_APP_ARG(durationst);
+		AST_APP_ARG(channel);
+	);
+
+	if (ast_strlen_zero(vdata)) {
+		ast_log(LOG_WARNING, "SendMF requires an argument\n");
+		return 0;
+	}
+
+	data = ast_strdupa(vdata);
+	AST_STANDARD_APP_ARGS(args, data);
+
+	if (ast_strlen_zero(args.digits)) {
+		ast_log(LOG_WARNING, "The digits argument is required (0-9,*#ABC,wf)\n");
+		return 0;
+	}
+	if (!ast_strlen_zero(args.dinterval)) {
+		ast_app_parse_timelen(args.dinterval, &dinterval, TIMELEN_MILLISECONDS);
+	}
+	if (!ast_strlen_zero(args.duration)) {
+		ast_app_parse_timelen(args.duration, &duration, TIMELEN_MILLISECONDS);
+	}
+	if (!ast_strlen_zero(args.durationkp)) {
+		ast_app_parse_timelen(args.durationkp, &durationkp, TIMELEN_MILLISECONDS);
+	}
+	if (!ast_strlen_zero(args.durationst)) {
+		ast_app_parse_timelen(args.durationst, &durationst, TIMELEN_MILLISECONDS);
+	}
+	if (!ast_strlen_zero(args.channel)) {
+		chan_found = ast_channel_get_by_name(args.channel);
+		if (!chan_found) {
+			ast_log(LOG_WARNING, "No such channel: %s\n", args.channel);
+			return 0;
+		}
+		chan_dest = chan_found;
+		if (chan_found != chan) {
+			chan_autoservice = chan;
+		}
+	}
+	res = ast_mf_stream(chan_dest, chan_autoservice, NULL, args.digits, dinterval <= 0 ? MF_BETWEEN_MS : dinterval,
+		duration <= 0 ? MF_DURATION : duration, durationkp <= 0 ? MF_KP_DURATION : durationkp,
+		durationst <= 0 ? MF_ST_DURATION : durationst, 0);
+	ast_channel_cleanup(chan_found);
+
+	return chan_autoservice ? 0 : res;
+}
+
+static int manager_play_mf(struct mansession *s, const struct message *m)
+{
+	const char *channel = astman_get_header(m, "Channel");
+	const char *digit = astman_get_header(m, "Digit");
+	const char *duration = astman_get_header(m, "Duration");
+	struct ast_channel *chan;
+	unsigned int duration_ms = MF_DURATION;
+
+	if (!(chan = ast_channel_get_by_name(channel))) {
+		astman_send_error(s, m, "Channel not found");
+		return 0;
+	}
+
+	if (ast_strlen_zero(digit)) {
+		astman_send_error(s, m, "No digit specified");
+		chan = ast_channel_unref(chan);
+		return 0;
+	}
+
+	/* Override default duration with KP or ST-specific default durations */
+	if (!strcmp(digit, "*"))
+		duration_ms = MF_KP_DURATION;
+
+	if (!strcmp(digit, "#") || !strcmp(digit, "A") || !strcmp(digit, "B") || !strcmp(digit, "C"))
+		duration_ms = MF_ST_DURATION;
+
+	if (!ast_strlen_zero(duration) && (sscanf(duration, "%30u", &duration_ms) != 1)) {
+		astman_send_error(s, m, "Could not convert Duration parameter");
+		chan = ast_channel_unref(chan);
+		return 0;
+	}
+
+	ast_mf_stream(chan, NULL, NULL, digit, 0, duration_ms, duration_ms, duration_ms, 1);
+
+	chan = ast_channel_unref(chan);
+
+	astman_send_ack(s, m, "MF successfully queued");
+
+	return 0;
+}
+
+static int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_application(readmf_name);
+	res |= ast_unregister_application(sendmf_name);
+	res |= ast_manager_unregister("PlayMF");
+
+	return res;
+}
+
+static int load_module(void)
+{
+	int res;
+
+	res = ast_register_application_xml(readmf_name, read_mf_exec);
+	res |= ast_register_application_xml(sendmf_name, sendmf_exec);
+	res |= ast_manager_register_xml("PlayMF", EVENT_FLAG_CALL, manager_play_mf);
+
+	return res;
+}
+
+AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "MF Sender and Receiver Applications");

apps/app_milliwatt.c

@@ -40,19 +40,29 @@
 /*** DOCUMENTATION
 	<application name="Milliwatt" language="en_US">
 		<synopsis>
-			Generate a Constant 1004Hz tone at 0dbm (mu-law).
+			Generates a 1004 Hz test tone at 0dbm (mu-law).
 		</synopsis>
 		<syntax>
 			<parameter name="options">
 				<optionlist>
+					<option name="m">
+						<para>Generate a 1004 Hz Milliwatt test tone at 0dbm, with a
+						1 second silent interval. This option must be specified
+						if you are using this for a milliwatt test line.</para>
+					</option>
 					<option name="o">
-						<para>Generate the tone at 1000Hz like previous version.</para>
+						<para>Generate a constant tone at 1000 Hz like previous version.</para>
 					</option>
 				</optionlist>
 			</parameter>
 		</syntax>
 		<description>
-			<para>Previous versions of this application generated the tone at 1000Hz.  If for
+			<para>Generates a 1004 Hz test tone.</para>
+			<para>By default, this application does not provide a Milliwatt test tone. It simply
+			plays a 1004 Hz tone, which is not suitable for performing a milliwatt test.
+			The <literal>m</literal> option should be used so that a real Milliwatt test tone
+			is provided. This will include a 1 second silent interval every 10 seconds.</para>
+			<para>Previous versions of this application generated a constant tone at 1000 Hz. If for
 			some reason you would prefer that behavior, supply the <literal>o</literal> option to get the
 			old behavior.</para>
 		</description>
@@ -155,8 +165,11 @@ static int milliwatt_exec(struct ast_channel *chan, const char *data)
 	if (!ast_strlen_zero(options) && strchr(options, 'o')) {
 		return old_milliwatt_exec(chan);
 	}
-
-	res = ast_playtones_start(chan, 23255, "1004/1000", 0);
+	if (!ast_strlen_zero(options) && strchr(options, 'm')) {
+		res = ast_playtones_start(chan, 23255, "1004/9000,0/1000", 0);
+	} else {
+		res = ast_playtones_start(chan, 23255, "1004/1000", 0);
+	}
 
 	while (!res) {
 		res = ast_safe_sleep(chan, 10000);