Update for 21.2.0

* Added checks for missing session, session->channel and rdata
  in stir_shaken_incoming_request.

* Added checks for missing session, session->channel and tdata
  in stir_shaken_outgoing_request.

Resolves: #645

.github/workflows/CherryPickTest.yml

@@ -131,7 +131,7 @@ jobs:
           github_token:      ${{secrets.GITHUB_TOKEN}}
           testsuite_repo:    ${{vars.TESTSUITE_REPO}}
           gatetest_group:    ${{matrix.group}}
-          gatetest_commands: ${{vars.GATETEST_COMMANDS}}
+          gatetest_command:  ${{ toJSON(fromJSON(vars.GATETEST_COMMANDS)[matrix.group]) }}
 
   CherryPickGateTests:
     needs: [ IdentifyBranches, CherryPickGateTestMatrix ]

.github/workflows/CreateDocs.yml

@@ -4,7 +4,7 @@ on:
     inputs:
       branches:
         description: "JSON array of branches: ['18','20'] (no spaces)"
-        required:    true
+        required:    false
         type:        string
   schedule:
 #           Times are UTC
@@ -14,15 +14,29 @@ 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-vars: ${{toJSON(inputs)}}
+          action-inputs: ${{toJSON(inputs)}}
+          action-vars: ${{ toJSON(steps.setup.outputs) }}
 
   CreateDocsScheduledMatrix:
     needs: [ CreateDocsDebug ]
@@ -73,7 +87,7 @@ jobs:
     strategy:
       fail-fast: false
       matrix:
-        branch: ${{ fromJSON(inputs.branches) }}
+        branch: ${{ fromJSON(vars.WIKIDOC_MANUAL_BRANCHES) }}
     runs-on: ubuntu-latest
     steps:
       - name: CreateDocs for ${{matrix.branch}}

.github/workflows/MergeApproved.yml

@@ -123,9 +123,6 @@ jobs:
   MergeAndCherryPick:
     needs: [ IdentifyBranches, PreMergeUnitTests ]
     if: success()
-    concurrency:
-      group: MergeAndCherryPick
-      cancel-in-progress: false
     runs-on: ubuntu-latest
     steps:
       - name: Start Merge
@@ -138,7 +135,7 @@ jobs:
 
       - name: Get Token needed to push cherry-picks
         id: get_workflow_token
-        uses: peter-murray/workflow-application-token-action@v1
+        uses: peter-murray/workflow-application-token-action@v2
         with:
           application_id: ${{secrets.ASTERISK_ORG_ACCESS_APP_ID}}
           application_private_key: ${{secrets.ASTERISK_ORG_ACCESS_APP_PRIV_KEY}}

.github/workflows/NightlyTests.yml

@@ -33,7 +33,7 @@ jobs:
           github_token:      ${{secrets.GITHUB_TOKEN}}
           testsuite_repo:    ${{vars.TESTSUITE_REPO}}
           gatetest_group:    ${{matrix.group}}
-          gatetest_commands: ${{vars.GATETEST_COMMANDS}}
+          gatetest_command:  ${{ toJSON(fromJSON(vars.GATETEST_COMMANDS)[matrix.group]) }}
 
   AsteriskNightlyTests:
     if: ${{ always() }}

.github/workflows/PROpenedOrUpdated.yml

@@ -1,184 +0,0 @@
-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: Job Start Delay
-        env:
-          JOB_START_DELAY_SEC:       ${{vars.PR_JOB_START_DELAY_SEC}}
-        run: |
-          # Give the user a chance to add their "cherry-pick-to" comments
-          sleep ${JOB_START_DELAY_SEC:-60}
-
-      - 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
-        env:
-          GITHUB_TOKEN: ${{steps.get_workflow_token.outputs.token}}
-          GH_TOKEN: ${{steps.get_workflow_token.outputs.token}}
-          CHERRY_PICK_REMINDER: ${{vars.CHERRY_PICK_REMINDER}}
-          BRANCHES_OUTPUT: ${{toJSON(steps.getbranches.outputs)}}
-          BRANCH_COUNT: ${{steps.getbranches.outputs.branch_count}}
-          FORCED_NONE:  ${{steps.getbranches.outputs.forced_none}}
-        run: |
-          # If the user already added "cherry-pick-to" comments
-          # we don't need to remind them.
-          ( $FORCED_NONE || [ $BRANCH_COUNT -gt 0 ] ) && { echo "No reminder needed." ; exit 0 ; }
-          IFS=$'; \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/PRSubmitActions.yml

@@ -0,0 +1,148 @@
+name: PRSubmitActions
+run-name: "PRSubmitActions: Test ${{github.event.action}}"
+on:
+  workflow_run:
+    workflows: [PRSubmitTests]
+    types:
+      - requested
+      - completed
+env:
+  ACTION:     ${{ github.event.action }}
+  CONCLUSION: ${{ github.event.workflow_run.conclusion }}
+  REPO:       ${{ github.repository }}
+
+jobs:
+  PRSubmitActions:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Get PR Number
+        id: getpr
+        uses: actions/github-script@v7
+        with:
+          retries: 5
+          script: |
+            let search = `repo:${context.repo.owner}/${context.repo.repo} ${context.payload.workflow_run.head_sha}`; 
+            let prs = await github.rest.search.issuesAndPullRequests({
+                 q: search,
+              });
+            if (prs.data.total_count == 0) {
+              core.setFailed(`Unable to get PR for ${context.payload.workflow_run.head_sha}`);
+              return;
+            }
+            let pr_number = prs.data.items[0].number;
+            core.setOutput('pr_number', pr_number);
+            return;
+
+      - name: Set Label
+        id: setlabel
+        uses: actions/github-script@v7
+        env:
+          PR_NUMBER:  ${{ steps.getpr.outputs.PR_NUMBER }}
+          LABEL_TIP:  ${{ vars.PR_SUBMIT_TESTING_IN_PROGRESS }}
+          LABEL_PASS: ${{ vars.PR_SUBMIT_TESTS_PASSED }}
+          LABEL_FAIL: ${{ vars.PR_SUBMIT_TESTS_FAILED }}
+        with:
+          retries: 5
+          script: |
+            let label;
+            if (process.env.ACTION === 'requested') {
+              label = process.env.LABEL_TIP;
+            } else {
+              if ( process.env.CONCLUSION === 'success' ) {
+                label = process.env.LABEL_PASS;
+              } else {
+                label = process.env.LABEL_FAIL;
+              }
+            }
+            core.info(`Setting label ${label}`);
+            github.rest.issues.setLabels({
+              issue_number: process.env.PR_NUMBER,
+              owner: context.repo.owner,
+              repo: context.repo.repo,
+              labels: [ label ]
+            });
+            return;
+
+      - name: Get cherry-pick branches
+        if: github.event.action == 'completed'
+        id: getbranches
+        uses: asterisk/asterisk-ci-actions/GetCherryPickBranchesFromPR@main
+        with:
+          repo:              ${{env.REPO}}
+          pr_number:         ${{steps.getpr.outputs.PR_NUMBER}}
+          cherry_pick_regex: ${{vars.CHERRY_PICK_REGEX}}
+          github_token:      ${{secrets.GITHUB_TOKEN}}
+
+      - name: Add cherry-pick reminder
+        if: github.event.action == 'completed'
+        uses: actions/github-script@v7
+        env:
+          PR_NUMBER:            ${{steps.getpr.outputs.PR_NUMBER}}
+          CHERRY_PICK_REMINDER: ${{vars.CHERRY_PICK_REMINDER}}
+          BRANCHES_OUTPUT:      ${{toJSON(steps.getbranches.outputs)}}
+          BRANCH_COUNT:         ${{steps.getbranches.outputs.branch_count}}
+          FORCED_NONE:          ${{steps.getbranches.outputs.forced_none}}
+        with:
+          retries: 5
+          script: |
+            if (process.env.FORCED_NONE === 'true' ||
+              process.env.BRANCH_COUNT > 0) {
+              core.info("No cherry-pick reminder needed.");
+              return;
+            }
+            let comments = await github.rest.issues.listComments({
+              issue_number: process.env.PR_NUMBER,
+              owner: context.repo.owner,
+              repo: context.repo.repo,
+            });
+            let found = false;
+            for (const c of comments.data) {
+              if (c.body.startsWith("<!--CPR-->")) {
+                found = true;
+                break;
+              }
+            }
+            if (found) {
+              core.info("Cherry-pick reminder already exists.");
+              return;
+            }
+            core.info("Adding cherry-pick reminder.");
+            await github.rest.issues.createComment({
+              issue_number: process.env.PR_NUMBER,
+              owner: context.repo.owner,
+              repo: context.repo.repo,
+              body: process.env.CHERRY_PICK_REMINDER
+            })
+            return;
+
+      - name: Add reviewers
+        if: github.event.action == 'completed'
+        uses: actions/github-script@v7
+        env:
+          PR_NUMBER: ${{steps.getpr.outputs.PR_NUMBER}}
+          REVIEWERS: ${{vars.PR_REVIEWERS}}
+        with:
+          retries: 5
+          script: |
+            let rs = JSON.parse(process.env.REVIEWERS);
+            let users = [];
+            let teams = [];
+            for (const r of rs) {
+              if (r.indexOf("/") > 0) {
+                teams.push(r.split('/')[1]);
+              } else {
+                users.push(r);
+              }
+            }
+            if (teams.length > 0 || users.length > 0) {
+              core.info(`Adding user reviewers ${users}`);
+              core.info(`Adding team reviewers ${teams}`);
+              await github.rest.pulls.requestReviewers({
+                pull_number: process.env.PR_NUMBER,
+                owner: context.repo.owner,
+                repo: context.repo.repo,
+                reviewers: users,
+                team_reviewers: teams
+              });
+            }
+            return;

.github/workflows/PRSubmitTests.yml

@@ -0,0 +1,114 @@
+name: PRSubmitTests
+run-name: "PR ${{github.event.number}} ${{github.event.action}} by ${{ github.actor }}"
+on:
+  pull_request:
+    types: [opened, reopened, synchronize]
+
+concurrency:
+  group: ${{github.workflow}}-${{github.event.number}}
+  cancel-in-progress: true
+
+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}}
+
+jobs:
+#
+# Pull requests created from forked respositories don't have access to
+# the "Action Variables" ('vars' context) so we need to retrieve control
+# data from an action.
+#
+  PRSGetControlData:
+    runs-on: ubuntu-latest
+    outputs:
+      control_data:  ${{ steps.setvars.outputs.control_data }}
+    steps:
+      - id: setvars
+        uses: asterisk/asterisk-ci-actions/GetRepoControlData@main
+        with:
+          repo: ${{ github.event.repository.name}}
+      - name: DumpEnvironment
+        uses: asterisk/asterisk-ci-actions/DumpEnvironmentAction@main
+        with:
+          action-inputs: ${{toJSON(inputs)}}
+          action-vars: ${{ toJSON(steps.setvars.outputs) }}
+
+  PRSUnitTests:
+    needs: PRSGetControlData
+    runs-on: ubuntu-latest
+    env:
+      UNITTEST_COMMAND: ${{ fromJSON(needs.PRSGetControlData.outputs.control_data).UNITTEST_COMMAND }}
+    steps:
+      - 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}}
+          unittest_command:  ${{env.UNITTEST_COMMAND}}
+
+  PRSGateTestMatrix:
+    runs-on: ubuntu-latest
+    needs: PRSGetControlData
+    continue-on-error: false
+    strategy:
+      fail-fast: false
+      matrix:
+        group: ${{ fromJSON(fromJSON(needs.PRSGetControlData.outputs.control_data).GATETEST_LIST) }}
+    env:
+      TESTSUITE_REPO:    "${{ fromJSON(needs.PRSGetControlData.outputs.control_data).TESTSUITE_REPO }}"
+      GATETEST_COMMANDS: "${{ fromJSON(needs.PRSGetControlData.outputs.control_data).GATETEST_COMMANDS }}"
+      GATETEST_COMMAND: "${{ toJSON(fromJSON(fromJSON(needs.PRSGetControlData.outputs.control_data).GATETEST_COMMANDS)[matrix.group]) }}"
+    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}}
+          testsuite_repo:    ${{env.TESTSUITE_REPO}}
+          gatetest_group:    ${{matrix.group}}
+          gatetest_command:  ${{env.GATETEST_COMMAND}}
+
+  PRSTestResults:
+    if: always()
+    runs-on: ubuntu-latest
+    needs: [PRSUnitTests,PRSGateTestMatrix]
+    steps:
+      - name: Check test matrix status
+        env:
+          RESULT_UNIT: ${{ needs.PRSUnitTests.result }}
+          RESULT_GATE: ${{ needs.PRSGateTestMatrix.result }}
+        run: |
+          declare -i rc=0
+          echo "all results: ${{ toJSON(needs.*.result) }}"
+          case $RESULT_UNIT in
+            success)
+              echo "::notice::Unit tests passed"
+              ;;
+            skipped)
+              echo "::error::Unit tests were skipped because of an earlier failure"
+              rc+=1
+              ;;
+            *)
+              echo "::error::One or more unit tests failed ($RESULT_UNIT)"
+              rc+=1
+          esac
+          case $RESULT_GATE in
+            success)
+              echo "::notice::Gate tests passed"
+              ;;
+            skipped)
+              echo "::error::Gate tests were skipped because of an earlier failure"
+              rc+=1
+              ;;
+            *)
+              echo "::error::One or more gate tests failed ($RESULT_GATE)"
+              rc+=1
+          esac
+          echo "::notice::Final result code: $rc"
+          exit $rc

.github/workflows/Releaser.yml

@@ -1,6 +1,5 @@
-# yaml-language-server: $schema=https://json.schemastore.org/github-workflow.json
-name: Asterisk Release
-run-name: ${{ github.actor }} is creating Asterisk release ${{inputs.new_version}}
+name: Releaser
+run-name: ${{ github.actor }} is creating ${{vars.PRODUCT_NAME}} release ${{inputs.new_version}}
 on:
   workflow_dispatch:
     inputs:
@@ -12,13 +11,6 @@ on:
             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
       is_security:
         description: |
           Security?
@@ -40,6 +32,12 @@ on:
         required: true
         type: boolean
         default: false
+      force_cherry_pick:
+        description: |
+          Force cherry-pick for non-RC1 releases? USE WITH CAUTION!
+        required: true
+        type: boolean
+        default: false
       push_release_branches:
         description: |
           Push release branches live?
@@ -70,31 +68,32 @@ jobs:
     runs-on: ubuntu-latest
     steps:
       - name: Run Releaser
-        uses: asterisk/asterisk-ci-actions/AsteriskReleaserComposite@main
+        uses: asterisk/asterisk-ci-actions/ReleaserComposite@main
         with:
+          product: ${{vars.PRODUCT_NAME}}
           is_security: ${{inputs.is_security}}
           advisories: ${{inputs.advisories}}
           is_hotfix: ${{inputs.is_hotfix}}
           new_version: ${{inputs.new_version}}
-#          start_version: ${{inputs.start_version}}
+          force_cherry_pick: ${{inputs.force_cherry_pick}}
           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}}
-          asterisk_mail_list_ga: ${{vars.ASTERISK_MAIL_LIST_GA}}
-          asterisk_mail_list_rc: ${{vars.ASTERISK_MAIL_LIST_RC}}
-          asterisk_mail_list_cert_ga: ${{vars.ASTERISK_MAIL_LIST_CERT_GA}}
-          asterisk_mail_list_cert_rc: ${{vars.ASTERISK_MAIL_LIST_CERT_RC}}
-          asterisk_mail_list_sec: ${{vars.ASTERISK_MAIL_LIST_SEC_ADV}}
-          sec_adv_url_base: ${{vars.ASTERISK_SEC_ADV_URL_BASE}}
+          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.ASTERISK_DEPLOY_SSH_PRIV_KEY}}
-          deploy_ssh_username: ${{secrets.ASTERISK_DEPLOY_SSH_USERNAME}}
-          deploy_host: ${{vars.ASTERISK_DEPLOY_HOST}}
-          deploy_dir: ${{vars.ASTERISK_DEPLOY_DIR}}
+          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}}

.version

@@ -0,0 +1 @@
+21.2.0

BUGS

@@ -10,7 +10,7 @@ For more information on using the bug tracker, or to
 learn how you can contribute by acting as a bug marshal
 please see:
 
-	https://wiki.asterisk.org/wiki/x/RgAtAQ
+	https://docs.asterisk.org/Asterisk-Community/Asterisk-Issue-Guidelines/
 
 If you would like to submit a feature request, please
 resist the temptation to post it to the bug tracker.

CHANGES.md

@@ -0,0 +1 @@
+ChangeLogs/ChangeLog-21.2.0.md

ChangeLogs/ChangeLog-21.0.0.md

@@ -0,0 +1,766 @@
+
+Change Log for Release asterisk-21.0.0
+========================================
+
+Links:
+----------------------------------------
+
+ - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)  
+ - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)  
+ - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)  
+ - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  
+
+Summary:
+----------------------------------------
+
+- Update master branch for Asterisk 21
+- translate.c: Prefer better codecs upon translate ties.
+- chan_skinny: Remove deprecated module.
+- app_osplookup: Remove deprecated module.
+- chan_mgcp: Remove deprecated module.
+- chan_alsa: Remove deprecated module.
+- pbx_builtins: Remove deprecated and defunct functionality.
+- chan_sip: Remove deprecated module.
+- app_cdr: Remove deprecated application and option.
+- app_macro: Remove deprecated module.
+- res_monitor: Remove deprecated module.
+- http.c: Minor simplification to HTTP status output.
+- app_osplookup: Remove obsolete sample config.
+- say.c: Fix French time playback. (#42)
+- core: Cleanup gerrit and JIRA references. (#58)
+- utils.h: Deprecate `ast_gethostbyname()`. (#79)
+- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
+- app_sla: Migrate SLA applications out of app_meetme.
+- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
+- .github: Update AsteriskReleaser for security releases
+- users.conf: Deprecate users.conf configuration.
+- Update version for Asterisk 21
+- Remove unneeded CHANGES and UPGRADE files
+- res_pjsip_pubsub: Add body_type to test_handler for unit tests
+- ari-stubs: Fix more local anchor references
+- ari-stubs: Fix more local anchor references
+- ari-stubs: Fix broken documentation anchors
+- res_pjsip_session: Send Session Interval too small response
+- .github: Update workflow-application-token-action to v2
+- app_dial: Fix infinite loop when sending digits.
+- app_voicemail: Fix for loop declarations
+- alembic: Fix quoting of the 100rel column
+- pbx.c: Fix gcc 12 compiler warning.
+- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
+- download_externals:  Fix a few version related issues
+- main/refer.c: Fix double free in refer_data_destructor + potential leak
+- sig_analog: Add Called Subscriber Held capability.
+- Revert "app_stack: Print proper exit location for PBXless channels."
+- install_prereq: Fix dependency install on aarch64.
+- res_pjsip.c: Set contact_user on incoming call local Contact header
+- extconfig: Allow explicit DB result set ordering to be disabled.
+- rest-api: Run make ari-stubs
+- res_pjsip_header_funcs: Make prefix argument optional.
+- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
+- manager: Tolerate stasis messages with no channel snapshot.
+- Remove unneeded CHANGES and UPGRADE files
+
+User Notes:
+----------------------------------------
+
+- ### sig_analog: Add Called Subscriber Held capability.
+  Called Subscriber Held is now supported for analog
+  FXS channels, using the calledsubscriberheld option. This allows
+  a station  user to go on hook when receiving an incoming call
+  and resume from another phone on the same line by going on hook,
+  without disconnecting the call.
+
+- ### res_pjsip_header_funcs: Make prefix argument optional.
+  The prefix argument to PJSIP_HEADERS is now
+  optional. If not specified, all header names will be
+  returned.
+
+- ### http.c: Minor simplification to HTTP status output.
+  For bound addresses, the HTTP status page now combines the bound
+  address and bound port in a single line. Additionally, the SSL bind
+  address has been renamed to TLS.
+
+
+Upgrade Notes:
+----------------------------------------
+
+- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
+  ast_gethostbyname() has been deprecated and will be removed
+  in Asterisk 23. New code should use `ast_sockaddr_resolve()` and
+  `ast_sockaddr_resolve_first_af()`.
+
+- ### app_sla: Migrate SLA applications out of app_meetme.
+  The SLAStation and SLATrunk applications have been moved
+  from app_meetme to app_sla. If you are using these applications and have
+  autoload=no, you will need to explicitly load this module in modules.conf.
+
+- ### users.conf: Deprecate users.conf configuration.
+  The users.conf config is now deprecated
+  and will be removed in a future version of Asterisk.
+
+- ### res_monitor: Remove deprecated module.
+  This module was deprecated in Asterisk 16
+  and is now being removed in accordance with
+  the Asterisk Module Deprecation policy.
+  This also removes the 'w' and 'W' options
+  for app_queue.
+  MixMonitor should be default and only option
+  for all settings that previously used either
+  Monitor or MixMonitor.
+
+- ### app_osplookup: Remove deprecated module.
+  This module was deprecated in Asterisk 19
+  and is now being removed in accordance with
+  the Asterisk Module Deprecation policy.
+
+- ### app_cdr: Remove deprecated application and option.
+  The previously deprecated NoCDR application has been removed.
+  Additionally, the previously deprecated 'e' option to the ResetCDR
+  application has been removed.
+
+- ### app_macro: Remove deprecated module.
+  This module was deprecated in Asterisk 16
+  and is now being removed in accordance with
+  the Asterisk Module Deprecation policy.
+  For most modules that interacted with app_macro,
+  this change is limited to no longer looking for
+  the current context from the macrocontext when set.
+  The following modules have additional impacts:
+  app_dial - no longer supports M^ connected/redirecting macro
+  app_minivm - samples written using macro will no longer work.
+  The sample needs to be re-written
+  app_queue - can no longer call a macro on the called party's
+  channel.  Use gosub which is currently supported
+  ccss - no callback macro, gosub only
+  app_voicemail - no macro support
+  channel  - remove macrocontext and priority, no connected
+  line or redirection macro options
+  options - stdexten is deprecated to gosub as the default
+  and only options
+  pbx - removed macrolock
+  pbx_dundi - no longer look for macro
+  snmp - removed macro context, exten, and priority
+
+- ### translate.c: Prefer better codecs upon translate ties.
+  When setting up translation between two codecs the quality was not taken into account,
+  resulting in suboptimal translation. The quality is now taken into account,
+  which can reduce the number of translation steps required, and improve the resulting quality.
+
+- ### chan_sip: Remove deprecated module.
+  This module was deprecated in Asterisk 17
+  and is now being removed in accordance with
+  the Asterisk Module Deprecation policy.
+
+- ### chan_alsa: Remove deprecated module.
+  This module was deprecated in Asterisk 19
+  and is now being removed in accordance with
+  the Asterisk Module Deprecation policy.
+
+- ### pbx_builtins: Remove deprecated and defunct functionality.
+  The previously deprecated ImportVar and SetAMAFlags
+  applications have now been removed.
+
+- ### chan_mgcp: Remove deprecated module.
+  This module was deprecated in Asterisk 19
+  and is now being removed in accordance with
+  the Asterisk Module Deprecation policy.
+
+- ### chan_skinny: Remove deprecated module.
+  This module was deprecated in Asterisk 19
+  and is now being removed in accordance with
+  the Asterisk Module Deprecation policy.
+
+
+Closed Issues:
+----------------------------------------
+
+  - #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms
+  - #39: [Bug]: Remove .gitreview from repository.
+  - #41: [Bug]: say.c Time announcement does not say o'clock for the French language
+  - #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
+  - #78: [improvement]: Deprecate ast_gethostbyname()
+  - #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
+  - #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
+  - #183: [deprecation]: Deprecate users.conf
+  - #226: [improvement]: Apply contact_user to incoming calls
+  - #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
+  - #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
+  - #253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs
+  - #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered"
+  - #263: [bug]: download_externals doesn't always handle versions correctly
+  - #267: [bug]: ari: refer with display_name key in request body leads to crash
+  - #274: [bug]: Syntax Error in SQL Code
+  - #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement'
+  - #277: [bug]: pbx.c: Compiler error with gcc 12.2
+  - #281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits
+  - #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in build_resource_tree
+
+Commits By Author:
+----------------------------------------
+
+- ### Asterisk Development Team (1):
+  - Update for 21.0.0-rc1
+
+- ### Bastian Triller (1):
+  - res_pjsip_session: Send Session Interval too small response
+
+- ### George Joseph (9):
+  - Remove unneeded CHANGES and UPGRADE files
+  - pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
+  - rest-api: Run make ari-stubs
+  - download_externals:  Fix a few version related issues
+  - alembic: Fix quoting of the 100rel column
+  - .github: Update workflow-application-token-action to v2
+  - ari-stubs: Fix broken documentation anchors
+  - ari-stubs: Fix more local anchor references
+  - ari-stubs: Fix more local anchor references
+
+- ### Jason D. McCormick (1):
+  - install_prereq: Fix dependency install on aarch64.
+
+- ### Joshua C. Colp (1):
+  - manager: Tolerate stasis messages with no channel snapshot.
+
+- ### Matthew Fredrickson (1):
+  - Revert "app_stack: Print proper exit location for PBXless channels."
+
+- ### Maximilian Fridrich (1):
+  - main/refer.c: Fix double free in refer_data_destructor + potential leak
+
+- ### Mike Bradeen (1):
+  - app_voicemail: Fix for loop declarations
+
+- ### MikeNaso (1):
+  - res_pjsip.c: Set contact_user on incoming call local Contact header
+
+- ### Naveen Albert (4):
+  - res_pjsip_header_funcs: Make prefix argument optional.
+  - sig_analog: Add Called Subscriber Held capability.
+  - pbx.c: Fix gcc 12 compiler warning.
+  - app_dial: Fix infinite loop when sending digits.
+
+- ### Sean Bright (1):
+  - extconfig: Allow explicit DB result set ordering to be disabled.
+
+- ### zhengsh (1):
+  - app_audiosocket: Fixed timeout with -1 to avoid busy loop.
+
+
+Detail:
+----------------------------------------
+
+- ### Update master branch for Asterisk 21
+  Author: George Joseph  
+  Date:   2022-07-20  
+
+
+- ### translate.c: Prefer better codecs upon translate ties.
+  Author: Naveen Albert  
+  Date:   2021-05-27  
+
+  If multiple codecs are available for the same
+  resource and the translation costs between
+  multiple codecs are the same, ties are
+  currently broken arbitrarily, which means a
+  lower quality codec would be used. This forces
+  Asterisk to explicitly use the higher quality
+  codec, ceteris paribus.
+
+  ASTERISK-29455
+
+
+- ### chan_skinny: Remove deprecated module.
+  Author: Mike Bradeen  
+  Date:   2022-11-16  
+
+  ASTERISK-30300
+
+
+- ### app_osplookup: Remove deprecated module.
+  Author: Mike Bradeen  
+  Date:   2022-11-18  
+
+  ASTERISK-30302
+
+
+- ### chan_mgcp: Remove deprecated module.
+  Author: Mike Bradeen  
+  Date:   2022-11-15  
+
+  Also removes res_pktcops to avoid merge conflicts
+  with ASTERISK~30301.
+
+  ASTERISK-30299
+
+
+- ### chan_alsa: Remove deprecated module.
+  Author: Mike Bradeen  
+  Date:   2022-11-14  
+
+  ASTERISK-30298
+
+
+- ### pbx_builtins: Remove deprecated and defunct functionality.
+  Author: Naveen Albert  
+  Date:   2022-11-29  
+
+  This removes the ImportVar and SetAMAFlags applications
+  which have been deprecated since Asterisk 12, but were
+  never removed previously.
+
+  Additionally, it removes remnants of defunct options
+  that themselves were removed years ago.
+
+  ASTERISK-30335 #close
+
+
+- ### chan_sip: Remove deprecated module.
+  Author: Mike Bradeen  
+  Date:   2022-11-28  
+
+  ASTERISK-30297
+
+
+- ### app_cdr: Remove deprecated application and option.
+  Author: Naveen Albert  
+  Date:   2022-12-22  
+
+  This removes the deprecated NoCDR application, which
+  was deprecated in Asterisk 12, having long been fully
+  superseded by the CDR_PROP function.
+
+  The deprecated e option to ResetCDR is also removed
+  for the same reason.
+
+  ASTERISK-30371 #close
+
+
+- ### app_macro: Remove deprecated module.
+  Author: Mike Bradeen  
+  Date:   2022-12-12  
+
+  For most modules that interacted with app_macro, this change is limited
+  to no longer looking for the current context from the macrocontext when
+  set.  Additionally, the following modules are impacted:
+
+  app_dial - no longer supports M^ connected/redirecting macro
+  app_minivm - samples written using macro will no longer work.
+  The sample needs a re-write
+
+  app_queue - can no longer a macro on the called party's channel.
+  Use gosub which is currently supported
+
+  ccss - no callback macro, gosub only
+
+  app_voicemail - no macro support
+
+  channel  - remove macrocontext and priority, no connected line or
+  redirection macro options
+  options - stdexten is deprecated to gosub as the default and only
+  pbx - removed macrolock
+  pbx_dundi - no longer look for macro
+
+  snmp - removed macro context, exten, and priority
+
+  ASTERISK-30304
+
+
+- ### res_monitor: Remove deprecated module.
+  Author: Mike Bradeen  
+  Date:   2022-11-18  
+
+  ASTERISK-30303
+
+
+- ### http.c: Minor simplification to HTTP status output.
+  Author: Boris P. Korzun  
+  Date:   2023-01-05  
+
+  Change the HTTP status page (located at /httpstatus by default) by:
+
+  * Combining the address and port into a single line.
+  * Changing "SSL" to "TLS"
+
+  ASTERISK-30433 #close
+
+
+- ### app_osplookup: Remove obsolete sample config.
+  Author: Naveen Albert  
+  Date:   2023-02-24  
+
+  ASTERISK_30302 previously removed app_osplookup,
+  but its sample config was not removed.
+  This removes it since nothing else uses it.
+
+  ASTERISK-30438 #close
+
+
+- ### say.c: Fix French time playback. (#42)
+  Author: InterLinked1  
+  Date:   2023-05-02  
+
+  ast_waitstream was not called after ast_streamfile,
+  resulting in "o'clock" being skipped in French.
+  
+  Additionally, the minute announcements should be
+  feminine.
+  
+  Reported-by: Danny Lloyd
+  
+  Resolves: #41
+  ASTERISK-30488
+- ### core: Cleanup gerrit and JIRA references. (#58)
+  Author: Sean Bright  
+  Date:   2023-05-03  
+
+  * Remove .gitreview and switch to pulling the main asterisk branch
+    version from configure.ac instead.
+  
+  * Replace references to JIRA with GitHub.
+  
+  * Other minor cleanup found along the way.
+  
+  Resolves: #39
+- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
+  Author: Sean Bright  
+  Date:   2023-05-11  
+
+  Deprecate `ast_gethostbyname()` in favor of `ast_sockaddr_resolve()` and
+  `ast_sockaddr_resolve_first_af()`. `ast_gethostbyname()` has not been
+  used by any in-tree code since 2021.
+  
+  This function will be removed entirely in Asterisk 23.
+  
+  Resolves: #78
+  
+  UpgradeNote: ast_gethostbyname() has been deprecated and will be removed
+  in Asterisk 23. New code should use `ast_sockaddr_resolve()` and
+  `ast_sockaddr_resolve_first_af()`.
+- ### res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
+  Author: InterLinked1  
+  Date:   2023-05-18  
+
+  The existing res_pjsip_pubsub APIs are somewhat limited in
+  what they can do. This adds a few API extensions that make
+  it possible for PJSIP pubsub modules to implement richer
+  features than is currently possible.
+  
+  * Allow pubsub modules to get a handle to pjsip_rx_data on subscription
+  * Allow pubsub modules to run a callback when a subscription is renewed
+  * Allow pubsub modules to run a callback for outgoing NOTIFYs, with
+    a handle to the tdata, so that modules can append their own headers
+    to the NOTIFYs
+  
+  This change does not add any features directly, but makes possible
+  several new features that will be added in future changes.
+  
+  Resolves: #81
+  ASTERISK-30485 #close
+  
+  Master-Only: True
+- ### app_sla: Migrate SLA applications out of app_meetme.
+  Author: Naveen Albert  
+  Date:   2023-05-02  
+
+  This removes the dependency of the SLAStation and SLATrunk
+  applications on app_meetme, in anticipation of the imminent
+  removal of the deprecated app_meetme module.
+
+  The user interface for the SLA applications is exactly the
+  same, and in theory, users should not notice a difference.
+  However, the SLA applications now use ConfBridge under the
+  hood, rather than MeetMe, and they are now contained within
+  their own module.
+
+  Resolves: #50
+  ASTERISK-30309
+
+  UpgradeNote: The SLAStation and SLATrunk applications have been moved
+  from app_meetme to app_sla. If you are using these applications and have
+  autoload=no, you will need to explicitly load this module in modules.conf.
+
+- ### rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
+  Author: George Joseph  
+  Date:   2023-06-27  
+
+
+- ### .github: Update AsteriskReleaser for security releases
+  Author: George Joseph  
+  Date:   2023-07-07  
+
+
+- ### users.conf: Deprecate users.conf configuration.
+  Author: Naveen Albert  
+  Date:   2023-06-30  
+
+  This deprecates the users.conf config file, which
+  is no longer as widely supported but still integrated
+  with a number of different modules.
+
+  Because there is no real mechanism for marking a
+  configuration file as "deprecated", and users.conf
+  is not just used in a single place, this now emits
+  a warning to the user when the PBX loads to notify
+  about the deprecation.
+
+  This configuration mechanism has been widely criticized
+  and discouraged since its inception, and is no longer
+  relevant to the configuration that most users are doing
+  today. Removing it will allow for some simplification
+  and cleanup in the codebase.
+
+  Resolves: #183
+
+  UpgradeNote: The users.conf config is now deprecated
+  and will be removed in a future version of Asterisk.
+
+- ### Update version for Asterisk 21
+  Author: George Joseph  
+  Date:   2023-08-09  
+
+
+- ### Remove unneeded CHANGES and UPGRADE files
+  Author: George Joseph  
+  Date:   2023-08-09  
+
+
+- ### res_pjsip_pubsub: Add body_type to test_handler for unit tests
+  Author: George Joseph  
+  Date:   2023-09-15  
+
+  The ast_sip_subscription_handler "test_handler" used for the unit
+  tests didn't set "body_type" so the NULL value was causing
+  a SEGV in build_subscription_tree().  It's now set to "".
+
+  Resolves: #335
+
+- ### ari-stubs: Fix more local anchor references
+  Author: George Joseph  
+  Date:   2023-09-05  
+
+  Also allow CreateDocs job to be run manually with default branches.
+
+
+- ### ari-stubs: Fix more local anchor references
+  Author: George Joseph  
+  Date:   2023-09-05  
+
+  Also allow CreateDocs job to be run manually with default branches.
+
+
+- ### ari-stubs: Fix broken documentation anchors
+  Author: George Joseph  
+  Date:   2023-09-05  
+
+  All of the links that reference page anchors with capital letters in
+  the ids (#Something) have been changed to lower case to match the
+  anchors that are generated by mkdocs.
+
+
+- ### res_pjsip_session: Send Session Interval too small response
+  Author: Bastian Triller  
+  Date:   2023-08-28  
+
+  Handle session interval lower than endpoint's configured minimum timer
+  when sending first answer. Timer setting is checked during this step and
+  needs to handled appropriately.
+  Before this change, no response was sent at all. After this change a
+  response with 422 Session Interval too small is sent to UAC.
+
+
+- ### .github: Update workflow-application-token-action to v2
+  Author: George Joseph  
+  Date:   2023-08-31  
+
+
+- ### app_dial: Fix infinite loop when sending digits.
+  Author: Naveen Albert  
+  Date:   2023-08-28  
+
+  If the called party hangs up while digits are being
+  sent, -1 is returned to indicate so, but app_dial
+  was not checking the return value, resulting in
+  the hangup being lost and looping forever until
+  the caller manually hangs up the channel. We now
+  abort if digit sending fails.
+
+  ASTERISK-29428 #close
+
+  Resolves: #281
+
+- ### app_voicemail: Fix for loop declarations
+  Author: Mike Bradeen  
+  Date:   2023-08-29  
+
+  Resolve for loop initial declarations added in cli changes.
+
+  Resolves: #275
+
+- ### alembic: Fix quoting of the 100rel column
+  Author: George Joseph  
+  Date:   2023-08-28  
+
+  Add quoting around the ps_endpoints 100rel column in the ALTER
+  statements.  Although alembic doesn't complain when generating
+  sql statements, postgresql does (rightly so).
+
+  Resolves: #274
+
+- ### pbx.c: Fix gcc 12 compiler warning.
+  Author: Naveen Albert  
+  Date:   2023-08-27  
+
+  Resolves: #277
+
+- ### app_audiosocket: Fixed timeout with -1 to avoid busy loop.
+  Author: zhengsh  
+  Date:   2023-08-24  
+
+  Resolves: asterisk#234
+
+- ### download_externals:  Fix a few version related issues
+  Author: George Joseph  
+  Date:   2023-08-18  
+
+  * Fixed issue with the script not parsing the new tag format for
+    certified releases.  The format changed from certified/18.9-cert5
+    to certified-18.9-cert5.
+
+  * Fixed issue where the asterisk version wasn't being considered
+    when looking for cached versions.
+
+  Resolves: #263
+
+- ### main/refer.c: Fix double free in refer_data_destructor + potential leak
+  Author: Maximilian Fridrich  
+  Date:   2023-08-21  
+
+  Resolves: #267
+
+- ### sig_analog: Add Called Subscriber Held capability.
+  Author: Naveen Albert  
+  Date:   2023-08-09  
+
+  This adds support for Called Subscriber Held for FXS
+  lines, which allows users to go on hook when receiving
+  a call and resume the call later from another phone on
+  the same line, without disconnecting the call. This is
+  a convenience mechanism that most real PSTN telephone
+  switches support.
+
+  ASTERISK-30372 #close
+
+  Resolves: #240
+
+  UserNote: Called Subscriber Held is now supported for analog
+  FXS channels, using the calledsubscriberheld option. This allows
+  a station  user to go on hook when receiving an incoming call
+  and resume from another phone on the same line by going on hook,
+  without disconnecting the call.
+
+
+- ### Revert "app_stack: Print proper exit location for PBXless channels."
+  Author: Matthew Fredrickson  
+  Date:   2023-08-10  
+
+  This reverts commit 617dad4cba1513dddce87b8e95a61415fb587cf1.
+
+  apps/app_stack.c: Revert buggy gosub patch
+
+  This seems to break the case when a predial macro calls a gosub.
+  When the gosub calls return, the Return function outputs:
+
+  app_stack.c:423 return_exec: Return without Gosub: stack is empty
+
+  This returns -1 to the calling macro, which returns to app_dial
+  and causes the call to hangup instead of proceeding with the macro
+  that invoked the gosub.
+
+  Resolves: #253
+
+- ### install_prereq: Fix dependency install on aarch64.
+  Author: Jason D. McCormick  
+  Date:   2023-04-28  
+
+  Fixes dependency solutions in install_prereq for Debian aarch64
+  platforms. install_prereq was attempting to forcibly install 32-bit
+  armhf packages due to the aptitude search for dependencies.
+
+  Resolves: #37
+
+- ### res_pjsip.c: Set contact_user on incoming call local Contact header
+  Author: MikeNaso  
+  Date:   2023-08-08  
+
+  If the contact_user is configured on the endpoint it will now be set on the local Contact header URI for incoming calls. The contact_user has already been set on the local Contact header URI for outgoing calls.
+
+  Resolves: #226
+
+- ### extconfig: Allow explicit DB result set ordering to be disabled.
+  Author: Sean Bright  
+  Date:   2023-07-12  
+
+  Added a new boolean configuration flag -
+  `order_multi_row_results_by_initial_column` - to both res_pgsql.conf
+  and res_config_odbc.conf that allows the administrator to disable the
+  explicit `ORDER BY` that was previously being added to all generated
+  SQL statements that returned multiple rows.
+
+  Fixes: #179
+
+- ### rest-api: Run make ari-stubs
+  Author: George Joseph  
+  Date:   2023-08-09  
+
+  An earlier cherry-pick that involved rest-api somehow didn't include
+  a comment change in res/ari/resource_endpoints.h.  This commit
+  corrects that.  No changes other than the comment.
+
+
+- ### res_pjsip_header_funcs: Make prefix argument optional.
+  Author: Naveen Albert  
+  Date:   2023-08-09  
+
+  The documentation for PJSIP_HEADERS claims that
+  prefix is optional, but in the code it is actually not.
+  However, there is no inherent reason for this, as users
+  may want to retrieve all header names, not just those
+  beginning with a certain prefix.
+
+  This makes the prefix optional for this function,
+  simply fetching all header names if not specified.
+  As a result, the documentation is now correct.
+
+  Resolves: #230
+
+  UserNote: The prefix argument to PJSIP_HEADERS is now
+  optional. If not specified, all header names will be
+  returned.
+
+
+- ### pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
+  Author: George Joseph  
+  Date:   2023-08-11  
+
+  The default is 32 with 8 being used by pjproject itself.  Recent
+  commits have put us over the limit resulting in assertions in
+  pjproject.  Since this value is used in invites, dialogs,
+  transports and subscriptions as well as the global pjproject
+  endpoint, we don't want to increase it too much.
+
+  Resolves: #255
+
+- ### manager: Tolerate stasis messages with no channel snapshot.
+  Author: Joshua C. Colp  
+  Date:   2023-08-09  
+
+  In some cases I have yet to determine some stasis messages may
+  be created without a channel snapshot. This change adds some
+  tolerance to this scenario, preventing a crash from occurring.
+
+
+- ### Remove unneeded CHANGES and UPGRADE files
+  Author: George Joseph  
+  Date:   2023-08-09  
+
+

ChangeLogs/ChangeLog-21.0.1.md

@@ -0,0 +1,172 @@
+
+Change Log for Release asterisk-21.0.1
+========================================
+
+Links:
+----------------------------------------
+
+ - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)  
+ - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)  
+ - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)  
+ - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  
+
+Summary:
+----------------------------------------
+
+- res_pjsip_header_funcs: Duplicate new header value, don't copy.
+- res_pjsip: disable raw bad packet logging
+- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
+- manager.c: Prevent path traversal with GetConfig.
+
+User Notes:
+----------------------------------------
+
+- ### http.c: Minor simplification to HTTP status output.
+  For bound addresses, the HTTP status page now combines the bound
+  address and bound port in a single line. Additionally, the SSL bind
+  address has been renamed to TLS.
+
+
+Upgrade Notes:
+----------------------------------------
+
+- ### chan_sip: Remove deprecated module.
+  This module was deprecated in Asterisk 17
+  and is now being removed in accordance with
+  the Asterisk Module Deprecation policy.
+
+- ### res_monitor: Remove deprecated module.
+  This module was deprecated in Asterisk 16
+  and is now being removed in accordance with
+  the Asterisk Module Deprecation policy.
+  This also removes the 'w' and 'W' options
+  for app_queue.
+  MixMonitor should be default and only option
+  for all settings that previously used either
+  Monitor or MixMonitor.
+
+- ### app_osplookup: Remove deprecated module.
+  This module was deprecated in Asterisk 19
+  and is now being removed in accordance with
+  the Asterisk Module Deprecation policy.
+
+- ### app_cdr: Remove deprecated application and option.
+  The previously deprecated NoCDR application has been removed.
+  Additionally, the previously deprecated 'e' option to the ResetCDR
+  application has been removed.
+
+- ### chan_skinny: Remove deprecated module.
+  This module was deprecated in Asterisk 19
+  and is now being removed in accordance with
+  the Asterisk Module Deprecation policy.
+
+- ### chan_mgcp: Remove deprecated module.
+  This module was deprecated in Asterisk 19
+  and is now being removed in accordance with
+  the Asterisk Module Deprecation policy.
+
+- ### translate.c: Prefer better codecs upon translate ties.
+  When setting up translation between two codecs the quality was not taken into account,
+  resulting in suboptimal translation. The quality is now taken into account,
+  which can reduce the number of translation steps required, and improve the resulting quality.
+
+- ### app_macro: Remove deprecated module.
+  This module was deprecated in Asterisk 16
+  and is now being removed in accordance with
+  the Asterisk Module Deprecation policy.
+  For most modules that interacted with app_macro,
+  this change is limited to no longer looking for
+  the current context from the macrocontext when set.
+  The following modules have additional impacts:
+  app_dial - no longer supports M^ connected/redirecting macro
+  app_minivm - samples written using macro will no longer work.
+  The sample needs to be re-written
+  app_queue - can no longer call a macro on the called party's
+  channel.  Use gosub which is currently supported
+  ccss - no callback macro, gosub only
+  app_voicemail - no macro support
+  channel  - remove macrocontext and priority, no connected
+  line or redirection macro options
+  options - stdexten is deprecated to gosub as the default
+  and only options
+  pbx - removed macrolock
+  pbx_dundi - no longer look for macro
+  snmp - removed macro context, exten, and priority
+
+- ### chan_alsa: Remove deprecated module.
+  This module was deprecated in Asterisk 19
+  and is now being removed in accordance with
+  the Asterisk Module Deprecation policy.
+
+- ### pbx_builtins: Remove deprecated and defunct functionality.
+  The previously deprecated ImportVar and SetAMAFlags
+  applications have now been removed.
+
+
+Closed Issues:
+----------------------------------------
+
+None
+
+Commits By Author:
+----------------------------------------
+
+- ### Ben Ford (1):
+  - manager.c: Prevent path traversal with GetConfig.
+
+- ### George Joseph (1):
+  - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
+
+- ### Gitea (1):
+  - res_pjsip_header_funcs: Duplicate new header value, don't copy.
+
+- ### Mike Bradeen (1):
+  - res_pjsip: disable raw bad packet logging
+
+
+Detail:
+----------------------------------------
+
+- ### res_pjsip_header_funcs: Duplicate new header value, don't copy.
+  Author: Gitea  
+  Date:   2023-07-10  
+
+  When updating an existing header the 'update' code incorrectly
+  just copied the new value into the existing buffer. If the
+  new value exceeded the available buffer size memory outside
+  of the buffer would be written into, potentially causing
+  a crash.
+
+  This change makes it so that the 'update' now duplicates
+  the new header value instead of copying it into the existing
+  buffer.
+
+- ### res_pjsip: disable raw bad packet logging
+  Author: Mike Bradeen  
+  Date:   2023-07-25  
+
+  Add patch to split the log level for invalid packets received on the
+  signaling port.  The warning regarding the packet will move to level 2
+  so that it can still be displayed, while the raw packet will be at level
+  4.
+
+- ### res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
+  Author: George Joseph  
+  Date:   2023-11-09  
+
+  When ICE is in use, we can prevent a possible DOS attack by allowing
+  DTLS protocol messages (client hello, etc) only from sources that
+  are in the active remote candidates list.
+
+  Resolves: GHSA-hxj9-xwr8-w8pq
+
+- ### manager.c: Prevent path traversal with GetConfig.
+  Author: Ben Ford  
+  Date:   2023-11-13  
+
+  When using AMI GetConfig, it was possible to access files outside of the
+  Asterisk configuration directory by using filenames with ".." and "./"
+  even while live_dangerously was not enabled. This change resolves the
+  full path and ensures we are still in the configuration directory before
+  attempting to access the file.
+

ChangeLogs/ChangeLog-21.0.2.md

@@ -0,0 +1,68 @@
+
+Change Log for Release asterisk-21.0.2
+========================================
+
+Links:
+----------------------------------------
+
+ - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)  
+ - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)  
+ - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)  
+ - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  
+
+Summary:
+----------------------------------------
+
+- res_rtp_asterisk: Fix regression issues with DTLS client check
+
+User Notes:
+----------------------------------------
+
+
+Upgrade Notes:
+----------------------------------------
+
+
+Closed Issues:
+----------------------------------------
+
+  - #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
+  - #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when it shouldn't
+  - #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr()
+
+Commits By Author:
+----------------------------------------
+
+- ### George Joseph (1):
+  - res_rtp_asterisk: Fix regression issues with DTLS client check
+
+
+Detail:
+----------------------------------------
+
+- ### res_rtp_asterisk: Fix regression issues with DTLS client check
+  Author: George Joseph  
+  Date:   2023-12-15  
+
+  * Since ICE candidates are used for the check and pjproject is
+    required to use ICE, res_rtp_asterisk was failing to compile
+    when pjproject wasn't available.  The check is now wrapped
+    with an #ifdef HAVE_PJPROJECT.
+
+  * The rtp->ice_active_remote_candidates container was being
+    used to check the address on incoming packets but that
+    container doesn't contain peer reflexive candidates discovered
+    during negotiation. This was causing the check to fail
+    where it shouldn't.  We now check against pjproject's
+    real_ice->rcand array which will contain those candidates.
+
+  * Also fixed a bug in ast_sockaddr_from_pj_sockaddr() where
+    we weren't zeroing out sin->sin_zero before returning.  This
+    was causing ast_sockaddr_cmp() to always return false when
+    one of the inputs was converted from a pj_sockaddr, even
+    if both inputs had the same address and port.
+
+  Resolves: #500
+  Resolves: #503
+  Resolves: #505
+

ChangeLogs/ChangeLog-21.2.0.md

@@ -0,0 +1,766 @@
+
+Change Log for Release asterisk-21.2.0
+========================================
+
+Links:
+----------------------------------------
+
+ - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)  
+ - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)  
+ - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)  
+ - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  
+
+Summary:
+----------------------------------------
+
+- res_pjsip_stir_shaken.c:  Add checks for missing parameters                     
+- app_dial: Add dial time for progress/ringing.                                   
+- app_voicemail: Properly reinitialize config after unit tests.                   
+- app_queue.c : fix "queue add member" usage string                               
+- app_voicemail: Allow preventing mark messages as urgent.                        
+- res_pjsip: Use consistent type for boolean columns.                             
+- attestation_config.c: Use ast_free instead of ast_std_free                      
+- Makefile: Add stir_shaken/cache to directories created on install               
+- Stir/Shaken Refactor                                                            
+- translate.c: implement new direct comp table mode                               
+- README.md: Removed outdated link                                                
+- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.              
+- res_rtp_asterisk.c: Correct coefficient in MOS calculation.                     
+- dsp.c: Fix and improve potentially inaccurate log message.                      
+- pjsip show channelstats: Prevent possible segfault when faxing                  
+- Reduce startup/shutdown verbose logging                                         
+- configure: Rerun bootstrap on modern platform.                                  
+- Upgrade bundled pjproject to 2.14.                                              
+- res_pjsip_outbound_registration.c: Add User-Agent header override               
+- app_speech_utils.c: Allow partial speech results.                               
+- utils: Make behavior of ast_strsep* match strsep.                               
+- app_chanspy: Add 'D' option for dual-channel audio                              
+- app_if: Fix next priority calculation.                                          
+- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.                          
+- BuildSystem: Bump autotools versions on OpenBSD.                                
+- main/utils: Simplify the FreeBSD ast_get_tid() handling                         
+- res_pjsip_session.c: Correctly format SDP connection addresses.                 
+- rtp_engine.c: Correct sample rate typo for L16/44100.                           
+- manager.c: Fix erroneous reloads in UpdateConfig.                               
+- res_calendar_icalendar: Print iCalendar error on parsing failure.               
+- app_confbridge: Don't emit warnings on valid configurations.                    
+- app_voicemail_odbc: remove macrocontext from voicemail_messages table           
+- chan_dahdi: Allow MWI to be manually toggled on channels.                       
+- chan_rtp.c: MulticastRTP missing refcount without codec option                  
+- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak                    
+- func_frame_trace: Add CLI command to dump frame queue.                          
+
+User Notes:
+----------------------------------------
+
+- ### app_dial: Add dial time for progress/ringing.                                   
+  The timeout argument to Dial now allows
+  specifying the maximum amount of time to dial if
+  early media is not received.
+
+- ### app_voicemail: Allow preventing mark messages as urgent.                        
+  The leaveurgent mailbox option can now be used to
+  control whether callers may leave messages marked as 'Urgent'.
+
+- ### Stir/Shaken Refactor                                                            
+  Asterisk's stir-shaken feature has been refactored to
+  correct interoperability, RFC compliance, and performance issues.
+  See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
+  information.
+
+- ### Upgrade bundled pjproject to 2.14.                                              
+  Bundled pjproject has been upgraded to 2.14. For more
+  information on what all is included in this change, check out the
+  pjproject Github page: https://github.com/pjsip/pjproject/releases
+
+- ### res_pjsip_outbound_registration.c: Add User-Agent header override               
+  PJSIP outbound registrations now support a per-registration
+  User-Agent header
+
+- ### app_speech_utils.c: Allow partial speech results.                               
+  The SpeechBackground dialplan application now supports a 'p'
+  option that will return partial results from speech engines that
+  provide them when a timeout occurs.
+
+- ### app_chanspy: Add 'D' option for dual-channel audio                              
+  The ChanSpy application now accepts the 'D' option which
+  will interleave the spied audio within the outgoing frames. The
+  purpose of this is to allow the audio to be read as a Dual channel
+  stream with separate incoming and outgoing audio. Setting both the
+  'o' option and the 'D' option and results in the 'D' option being
+  ignored.
+
+- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table           
+  The fix requires removing the macrocontext column from the
+  voicemail_messages table in the voicemail database via alembic upgrade.
+
+- ### chan_dahdi: Allow MWI to be manually toggled on channels.                       
+  The 'dahdi set mwi' now allows MWI on channels
+  to be manually toggled if needed for troubleshooting.
+  Resolves: #440
+
+
+Upgrade Notes:
+----------------------------------------
+
+- ### Stir/Shaken Refactor                                                            
+  The stir-shaken refactor is a breaking change but since
+  it's not working now we don't think it matters. The
+  stir_shaken.conf file has changed significantly which means that
+  existing ones WILL need to be changed.  The stir_shaken.conf.sample
+  file in configs/samples/ has quite a bit more information.  This is
+  also an ABI breaking change since some of the existing objects
+  needed to be changed or removed, and new ones added.  Additionally,
+  if res_stir_shaken is enabled in menuselect, you'll need to either
+  have the development package for libjwt v1.15.3 installed or use
+  the --with-libjwt-bundled option with ./configure.
+
+- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table           
+  The fix requires that the voicemail database be upgraded via
+  alembic. Upgrading to the latest voicemail database via alembic will
+  remove the macrocontext column from the voicemail_messages table.
+
+
+Closed Issues:
+----------------------------------------
+
+  - #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
+  - #351: [improvement]: Refactor res_stir_shaken to use libjwt
+  - #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
+  - #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
+  - #492: [improvement]: res_calendar_icalendar: Print icalendar error if available on parsing failure
+  - #515: [improvement]: Implement option to override User-Agent-Header on a per-registration basis
+  - #527: [bug]: app_voicemail_odbc no longer working after removal of macrocontext.
+  - #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed assertion bad magic number 0x0 for object" after ~30 calls
+  - #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of channel frame queue
+  - #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
+  - #560: [bug]: EndIf() causes next priority to be skipped
+  - #565: [bug]: Application Read() returns immediately
+  - #569: [improvement]: Add option to interleave input and output frames on spied channel
+  - #572: [improvement]: Copy partial speech results when Asterisk is ready to move on but the speech backend is not
+  - #582: [improvement]: Reduce unneeded logging during startup and shutdown
+  - #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in older gcc versions
+  - #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not received
+  - #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault when a fax session is active
+  - #595: [improvement]: dsp.c: Fix and improve confusing warning message.
+  - #597: [bug]: wrong MOS calculation
+  - #601: [new-feature]: translate.c: implement new direct comp table mode (PR #585)
+  - #619: [new-feature]: app_voicemail: Allow preventing callers from marking messages as urgent
+  - #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
+  - #634: [bug]: make install doesn't create the stir_shaken cache directory
+  - #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
+  - #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c
+
+Commits By Author:
+----------------------------------------
+
+- ### Ben Ford (1):
+  - Upgrade bundled pjproject to 2.14.
+
+- ### Brad Smith (2):
+  - main/utils: Simplify the FreeBSD ast_get_tid() handling
+  - BuildSystem: Bump autotools versions on OpenBSD.
+
+- ### Flole998 (1):
+  - res_pjsip_outbound_registration.c: Add User-Agent header override
+
+- ### George Joseph (6):
+  - Reduce startup/shutdown verbose logging
+  - pjsip show channelstats: Prevent possible segfault when faxing
+  - Stir/Shaken Refactor
+  - Makefile: Add stir_shaken/cache to directories created on install
+  - attestation_config.c: Use ast_free instead of ast_std_free
+  - res_pjsip_stir_shaken.c:  Add checks for missing parameters
+
+- ### Joshua C. Colp (1):
+  - utils: Make behavior of ast_strsep* match strsep.
+
+- ### Mike Bradeen (2):
+  - app_voicemail_odbc: remove macrocontext from voicemail_messages table
+  - app_chanspy: Add 'D' option for dual-channel audio
+
+- ### Naveen Albert (10):
+  - func_frame_trace: Add CLI command to dump frame queue.
+  - chan_dahdi: Allow MWI to be manually toggled on channels.
+  - res_calendar_icalendar: Print iCalendar error on parsing failure.
+  - manager.c: Fix erroneous reloads in UpdateConfig.
+  - app_if: Fix next priority calculation.
+  - configure: Rerun bootstrap on modern platform.
+  - dsp.c: Fix and improve potentially inaccurate log message.
+  - app_voicemail: Allow preventing mark messages as urgent.
+  - app_voicemail: Properly reinitialize config after unit tests.
+  - app_dial: Add dial time for progress/ringing.
+
+- ### PeterHolik (2):
+  - chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
+  - chan_rtp.c: MulticastRTP missing refcount without codec option
+
+- ### Sean Bright (6):
+  - app_confbridge: Don't emit warnings on valid configurations.
+  - rtp_engine.c: Correct sample rate typo for L16/44100.
+  - res_pjsip_session.c: Correctly format SDP connection addresses.
+  - res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
+  - strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
+  - res_pjsip: Use consistent type for boolean columns.
+
+- ### Sebastian Jennen (1):
+  - translate.c: implement new direct comp table mode
+
+- ### Shaaah (1):
+  - app_queue.c : fix "queue add member" usage string
+
+- ### Shyju Kanaprath (1):
+  - README.md: Removed outdated link
+
+- ### cmaj (1):
+  - app_speech_utils.c: Allow partial speech results.
+
+- ### romryz (1):
+  - res_rtp_asterisk.c: Correct coefficient in MOS calculation.
+
+
+Detail:
+----------------------------------------
+
+- ### res_pjsip_stir_shaken.c:  Add checks for missing parameters                     
+  Author: George Joseph  
+  Date:   2024-03-11  
+
+  * Added checks for missing session, session->channel and rdata
+    in stir_shaken_incoming_request.
+
+  * Added checks for missing session, session->channel and tdata
+    in stir_shaken_outgoing_request.
+
+  Resolves: #645
+
+- ### app_dial: Add dial time for progress/ringing.                                   
+  Author: Naveen Albert  
+  Date:   2024-02-08  
+
+  Add a timeout option to control the amount of time
+  to wait if no early media is received before giving
+  up. This allows aborting early if the destination
+  is not being responsive.
+
+  Resolves: #588
+
+  UserNote: The timeout argument to Dial now allows
+  specifying the maximum amount of time to dial if
+  early media is not received.
+
+
+- ### app_voicemail: Properly reinitialize config after unit tests.                   
+  Author: Naveen Albert  
+  Date:   2024-02-29  
+
+  Most app_voicemail unit tests were not properly cleaning up
+  after themselves after running. This led to test mailboxes
+  lingering around in the system. It also meant that if any
+  unit tests in app_voicemail that create mailboxes were executed
+  and the module was not unloaded/loaded again prior to running
+  the test_voicemail_vm_info unit test, Asterisk would segfault
+  due to an attempt to copy a NULL string.
+
+  The load_config test did actually have logic to reinitialize
+  the config after the test. However, this did not work in practice
+  since load_config() would not reload the config since voicemail.conf
+  had not changed during the test; thus, additional logic has been
+  added to ensure that voicemail.conf is truly reloaded, after any
+  unit tests which modify the users list.
+
+  This prevents the SEGV due to invalid mailboxes lingering around,
+  and also ensures that the system state is restored to what it was
+  prior to the tests running.
+
+  Resolves: #629
+
+- ### app_queue.c : fix "queue add member" usage string                               
+  Author: Shaaah  
+  Date:   2024-01-23  
+
+  Fixing bracket placement in the "queue add member" cli usage string.
+
+
+- ### app_voicemail: Allow preventing mark messages as urgent.                        
+  Author: Naveen Albert  
+  Date:   2024-02-24  
+
+  This adds an option to allow preventing callers from leaving
+  messages marked as 'urgent'.
+
+  Resolves: #619
+
+  UserNote: The leaveurgent mailbox option can now be used to
+  control whether callers may leave messages marked as 'Urgent'.
+
+
+- ### res_pjsip: Use consistent type for boolean columns.                             
+  Author: Sean Bright  
+  Date:   2024-02-27  
+
+  This migrates the relevant schema objects from the `('yes', 'no')`
+  definition to the `('0', '1', 'off', 'on', 'false', 'true', 'yes', 'no')`
+  one.
+
+  Fixes #617
+
+
+- ### attestation_config.c: Use ast_free instead of ast_std_free                      
+  Author: George Joseph  
+  Date:   2024-03-05  
+
+  In as_check_common_config, we were calling ast_std_free on
+  raw_key but raw_key was allocated with ast_malloc so it
+  should be freed with ast_free.
+
+  Resolves: #636
+
+- ### Makefile: Add stir_shaken/cache to directories created on install               
+  Author: George Joseph  
+  Date:   2024-03-04  
+
+  The default location for the stir_shaken cache is
+  /var/lib/asterisk/keys/stir_shaken/cache but we were only creating
+  /var/lib/asterisk/keys/stir_shaken on istall.  We now create
+  the cache sub-directory.
+
+  Resolves: #634
+
+- ### Stir/Shaken Refactor                                                            
+  Author: George Joseph  
+  Date:   2023-10-26  
+
+  Why do we need a refactor?
+
+  The original stir/shaken implementation was started over 3 years ago
+  when little was understood about practical implementation.  The
+  result was an implementation that wouldn't actually interoperate
+  with any other stir-shaken implementations.
+
+  There were also a number of stir-shaken features and RFC
+  requirements that were never implemented such as TNAuthList
+  certificate validation, sending Reason headers in SIP responses
+  when verification failed but we wished to continue the call, and
+  the ability to send Media Key(mky) grants in the Identity header
+  when the call involved DTLS.
+
+  Finally, there were some performance concerns around outgoing
+  calls and selection of the correct certificate and private key.
+  The configuration was keyed by an arbitrary name which meant that
+  for every outgoing call, we had to scan the entire list of
+  configured TNs to find the correct cert to use.  With only a few
+  TNs configured, this wasn't an issue but if you have a thousand,
+  it could be.
+
+  What's changed?
+
+  * Configuration objects have been refactored to be clearer about
+    their uses and to fix issues.
+      * The "general" object was renamed to "verification" since it
+        contains parameters specific to the incoming verification
+        process.  It also never handled ca_path and crl_path
+        correctly.
+      * A new "attestation" object was added that controls the
+        outgoing attestation process.  It sets default certificates,
+        keys, etc.
+      * The "certificate" object was renamed to "tn" and had it's key
+        change to telephone number since outgoing call attestation
+        needs to look up certificates by telephone number.
+      * The "profile" object had more parameters added to it that can
+        override default parameters specified in the "attestation"
+        and "verification" objects.
+      * The "store" object was removed altogther as it was never
+        implemented.
+
+  * We now use libjwt to create outgoing Identity headers and to
+    parse and validate signatures on incoming Identiy headers.  Our
+    previous custom implementation was much of the source of the
+    interoperability issues.
+
+  * General code cleanup and refactor.
+      * Moved things to better places.
+      * Separated some of the complex functions to smaller ones.
+      * Using context objects rather than passing tons of parameters
+        in function calls.
+      * Removed some complexity and unneeded encapsuation from the
+        config objects.
+
+  Resolves: #351
+  Resolves: #46
+
+  UserNote: Asterisk's stir-shaken feature has been refactored to
+  correct interoperability, RFC compliance, and performance issues.
+  See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
+  information.
+
+  UpgradeNote: The stir-shaken refactor is a breaking change but since
+  it's not working now we don't think it matters. The
+  stir_shaken.conf file has changed significantly which means that
+  existing ones WILL need to be changed.  The stir_shaken.conf.sample
+  file in configs/samples/ has quite a bit more information.  This is
+  also an ABI breaking change since some of the existing objects
+  needed to be changed or removed, and new ones added.  Additionally,
+  if res_stir_shaken is enabled in menuselect, you'll need to either
+  have the development package for libjwt v1.15.3 installed or use
+  the --with-libjwt-bundled option with ./configure.
+
+
+- ### translate.c: implement new direct comp table mode                               
+  Author: Sebastian Jennen  
+  Date:   2024-02-25  
+
+  The new mode lists for each codec translation the actual real cost in cpu microseconds per second translated audio.
+  This allows to compare the real cpu usage of translations and helps in evaluation of codec implementation changes regarding performance (regression testing).
+
+  - add new table mode
+  - hide the 999999 comp values, as these only indicate an issue with transcoding
+  - hide the 0 values, as these also do not contain any information (only indicate a multistep transcoding)
+
+  Resolves: #601
+
+- ### README.md: Removed outdated link                                                
+  Author: Shyju Kanaprath  
+  Date:   2024-02-23  
+
+  Removed outdated link http://www.quicknet.net from README.md
+
+  cherry-pick-to: 18
+  cherry-pick-to: 20
+  cherry-pick-to: 21
+
+- ### strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.              
+  Author: Sean Bright  
+  Date:   2024-02-17  
+
+  If a dynamic string is created with an initial length of 0,
+  `ast_str_buffer(…)` will return an invalid pointer.
+
+  This was a secondary discovery when fixing #65.
+
+
+- ### res_rtp_asterisk.c: Correct coefficient in MOS calculation.                     
+  Author: romryz  
+  Date:   2024-02-06  
+
+  Media Experience Score relies on incorrect pseudo_mos variable
+  calculation. According to forming an opinion section of the
+  documentation, calculation relies on ITU-T G.107 standard:
+
+      https://docs.asterisk.org/Deployment/Media-Experience-Score/#forming-an-opinion
+
+  ITU-T G.107 Annex B suggests to calculate MOS with a coefficient
+  "seven times ten to the power of negative six", 7 * 10^(-6). which
+  would mean 6 digits after the decimal point. Current implementation
+  has 7 digits after the decimal point, which downrates the calls.
+
+  Fixes: #597
+
+- ### dsp.c: Fix and improve potentially inaccurate log message.                      
+  Author: Naveen Albert  
+  Date:   2024-02-09  
+
+  If ast_dsp_process is called with a codec besides slin, ulaw,
+  or alaw, a warning is logged that in-band DTMF is not supported,
+  but this message is not always appropriate or correct, because
+  ast_dsp_process is much more generic than just DTMF detection.
+
+  This logs a more generic message in those cases, and also improves
+  codec-mismatch logging throughout dsp.c by ensuring incompatible
+  codecs are printed out.
+
+  Resolves: #595
+
+- ### pjsip show channelstats: Prevent possible segfault when faxing                  
+  Author: George Joseph  
+  Date:   2024-02-09  
+
+  Under rare circumstances, it's possible for the original audio
+  session in the active_media_state default_session to be corrupted
+  instead of removed when switching to the t38/image media session
+  during fax negotiation.  This can cause a segfault when a "pjsip
+  show channelstats" attempts to print that audio media session's
+  rtp statistics.  In these cases, the active_media_state
+  topology is correctly showing only a single t38/image stream
+  so we now check that there's an audio stream in the topology
+  before attempting to use the audio media session to get the rtp
+  statistics.
+
+  Resolves: #592
+
+- ### Reduce startup/shutdown verbose logging                                         
+  Author: George Joseph  
+  Date:   2024-01-31  
+
+  When started with a verbose level of 3, asterisk can emit over 1500
+  verbose message that serve no real purpose other than to fill up
+  logs. When asterisk shuts down, it emits another 1100 that are of
+  even less use. Since the testsuite runs asterisk with a verbose
+  level of 3, and asterisk starts and stops for every one of the 700+
+  tests, the number of log messages is staggering.  Besides taking up
+  resources, it also makes it hard to debug failing tests.
+
+  This commit changes the log level for those verbose messages to 5
+  instead of 3 which reduces the number of log messages to only a
+  handful. Of course, NOTICE, WARNING and ERROR message are
+  unaffected.
+
+  There's also one other minor change...
+  ast_context_remove_extension_callerid2() logs a DEBUG message
+  instead of an ERROR if the extension you're deleting doesn't exist.
+  The pjsip_config_wizard calls that function to clean up the config
+  and has been triggering that annoying error message for years.
+
+  Resolves: #582
+
+- ### configure: Rerun bootstrap on modern platform.                                  
+  Author: Naveen Albert  
+  Date:   2024-02-12  
+
+  The last time configure was run, it was run on a system that
+  did not enable -std=gnu11 by default, which meant that the
+  restrict qualifier would not be recognized on certain platforms.
+  This regenerates the configure files from running bootstrap.sh,
+  so that these should be recognized on all supported platforms.
+
+  Resolves: #586
+
+- ### Upgrade bundled pjproject to 2.14.                                              
+  Author: Ben Ford  
+  Date:   2024-02-05  
+
+  Fixes: #406
+
+  UserNote: Bundled pjproject has been upgraded to 2.14. For more
+  information on what all is included in this change, check out the
+  pjproject Github page: https://github.com/pjsip/pjproject/releases
+
+
+- ### res_pjsip_outbound_registration.c: Add User-Agent header override               
+  Author: Flole998  
+  Date:   2023-12-13  
+
+  This introduces a setting for outbound registrations to override the
+  global User-Agent header setting.
+
+  Resolves: #515
+
+  UserNote: PJSIP outbound registrations now support a per-registration
+  User-Agent header
+
+
+- ### app_speech_utils.c: Allow partial speech results.                               
+  Author: cmaj  
+  Date:   2024-02-02  
+
+  Adds 'p' option to SpeechBackground() application.
+  With this option, when the app timeout is reached,
+  whatever the backend speech engine collected will
+  be returned as if it were the final, full result.
+  (This works for engines that make partial results.)
+
+  Resolves: #572
+
+  UserNote: The SpeechBackground dialplan application now supports a 'p'
+  option that will return partial results from speech engines that
+  provide them when a timeout occurs.
+
+
+- ### utils: Make behavior of ast_strsep* match strsep.                               
+  Author: Joshua C. Colp  
+  Date:   2024-01-31  
+
+  Given the scenario of passing an empty string to the
+  ast_strsep functions the functions would return NULL
+  instead of an empty string. This is counter to how
+  strsep itself works.
+
+  This change alters the behavior of the functions to
+  match that of strsep.
+
+  Fixes: #565
+
+- ### app_chanspy: Add 'D' option for dual-channel audio                              
+  Author: Mike Bradeen  
+  Date:   2024-01-31  
+
+  Adds the 'D' option to app chanspy that causes the input and output
+  frames of the spied channel to be interleaved in the spy output frame.
+  This allows the input and output of the spied channel to be decoded
+  separately by the receiver.
+
+  If the 'o' option is also set, the 'D' option is ignored as the
+  audio being spied is inherently one direction.
+
+  Fixes: #569
+
+  UserNote: The ChanSpy application now accepts the 'D' option which
+  will interleave the spied audio within the outgoing frames. The
+  purpose of this is to allow the audio to be read as a Dual channel
+  stream with separate incoming and outgoing audio. Setting both the
+  'o' option and the 'D' option and results in the 'D' option being
+  ignored.
+
+
+- ### app_if: Fix next priority calculation.                                          
+  Author: Naveen Albert  
+  Date:   2024-01-28  
+
+  Commit fa3922a4d28860d415614347d9f06c233d2beb07 fixed
+  a branching issue but "overshoots" when calculating
+  the next priority. This fixes that; accompanying
+  test suite tests have also been extended.
+
+  Resolves: #560
+
+- ### res_pjsip_t38.c: Permit IPv6 SDP connection addresses.                          
+  Author: Sean Bright  
+  Date:   2024-01-29  
+
+  The existing code prevented IPv6 addresses from being properly parsed.
+
+  Fixes #558
+
+
+- ### BuildSystem: Bump autotools versions on OpenBSD.                                
+  Author: Brad Smith  
+  Date:   2024-01-27  
+
+  Bump up to the more commonly used and modern versions of
+  autoconf and automake.
+
+
+- ### main/utils: Simplify the FreeBSD ast_get_tid() handling                         
+  Author: Brad Smith  
+  Date:   2024-01-27  
+
+  FreeBSD has had kernel threads for 20+ years.
+
+
+- ### res_pjsip_session.c: Correctly format SDP connection addresses.                 
+  Author: Sean Bright  
+  Date:   2024-01-27  
+
+  Resolves a regression identified by @justinludwig involving the
+  rendering of IPv6 addresses in outgoing SDP.
+
+  Also updates `media_address` on PJSIP endpoints so that if we are able
+  to parse the configured value as an IP we store it in a format that we
+  can directly use later. Based on my reading of the code it appeared
+  that one could configure `media_address` as:
+
+  ```
+  [foo]
+  type = endpoint
+  ...
+  media_address = [2001:db8::]
+  ```
+
+  And that value would be blindly copied into the outgoing SDP without
+  regard to its format.
+
+  Fixes #541
+
+
+- ### rtp_engine.c: Correct sample rate typo for L16/44100.                           
+  Author: Sean Bright  
+  Date:   2024-01-28  
+
+  Fixes #555
+
+
+- ### manager.c: Fix erroneous reloads in UpdateConfig.                               
+  Author: Naveen Albert  
+  Date:   2024-01-25  
+
+  Currently, a reload will always occur if the
+  Reload header is provided for the UpdateConfig
+  action. However, we should not be doing a reload
+  if the header value has a falsy value, per the
+  documentation, so this makes the reload behavior
+  consistent with the existing documentation.
+
+  Resolves: #551
+
+- ### res_calendar_icalendar: Print iCalendar error on parsing failure.               
+  Author: Naveen Albert  
+  Date:   2023-12-14  
+
+  If libical fails to parse a calendar, print the error message it provdes.
+
+  Resolves: #492
+
+- ### app_confbridge: Don't emit warnings on valid configurations.                    
+  Author: Sean Bright  
+  Date:   2024-01-21  
+
+  The numeric bridge profile options `internal_sample_rate` and
+  `maximum_sample_rate` are documented to accept the special values
+  `auto` and `none`, respectively. While these values currently work,
+  they also emit warnings when used which could be confusing for users.
+
+  In passing, also ensure that we only accept the documented range of
+  sample rate values between 8000 and 192000.
+
+  Fixes #546
+
+
+- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table           
+  Author: Mike Bradeen  
+  Date:   2024-01-10  
+
+  When app_macro was deprecated, the macrocontext column was removed from
+  the INSERT statement but the binds were not renumbered. This broke the
+  insert.
+
+  This change removes the macrocontext column via alembic and re-numbers
+  the existing columns in the INSERT.
+
+  Fixes: #527
+
+  UserNote: The fix requires removing the macrocontext column from the
+  voicemail_messages table in the voicemail database via alembic upgrade.
+
+  UpgradeNote: The fix requires that the voicemail database be upgraded via
+  alembic. Upgrading to the latest voicemail database via alembic will
+  remove the macrocontext column from the voicemail_messages table.
+
+
+- ### chan_dahdi: Allow MWI to be manually toggled on channels.                       
+  Author: Naveen Albert  
+  Date:   2023-11-10  
+
+  This adds a CLI command to manually toggle the MWI status
+  of a channel, useful for troubleshooting or resetting
+  MWI devices, similar to the capabilities offered with
+  SIP messaging to manually control MWI status.
+
+  UserNote: The 'dahdi set mwi' now allows MWI on channels
+  to be manually toggled if needed for troubleshooting.
+
+  Resolves: #440
+
+- ### chan_rtp.c: MulticastRTP missing refcount without codec option                  
+  Author: PeterHolik  
+  Date:   2024-01-15  
+
+  Fixes: #529
+
+- ### chan_rtp.c: Change MulticastRTP nameing to avoid memory leak                    
+  Author: PeterHolik  
+  Date:   2024-01-16  
+
+  Fixes: asterisk#536
+
+- ### func_frame_trace: Add CLI command to dump frame queue.                          
+  Author: Naveen Albert  
+  Date:   2024-01-12  
+
+  This adds a simple CLI command that can be used for
+  analyzing all frames currently queued to a channel.
+
+  A couple log messages are also adjusted to be more
+  useful in tracing bridging problems.
+
+  Resolves: #533
+

Makefile

@@ -377,7 +377,7 @@ $(MOD_SUBDIRS_MENUSELECT_TREE):
 	+@$(SUBMAKE) -C $(@:-menuselect-tree=) SUBDIR=$(@:-menuselect-tree=) moduleinfo
 	+@$(SUBMAKE) -C $(@:-menuselect-tree=) SUBDIR=$(@:-menuselect-tree=) makeopts
 
-$(SUBDIRS): makeopts .lastclean main/version.c include/asterisk/build.h include/asterisk/buildopts.h defaults.h
+$(SUBDIRS): makeopts .lastclean main/version.c include/asterisk/build.h defaults.h
 
 ifeq ($(findstring $(OSARCH), mingw32 cygwin ),)
 main: third-party
@@ -403,7 +403,7 @@ defaults.h: makeopts .lastclean build_tools/make_defaults_h
 	@cmp -s $@.tmp $@ || mv $@.tmp $@
 	@rm -f $@.tmp
 
-main/version.c: FORCE menuselect.makeopts .lastclean
+main/version.c: FORCE include/asterisk/buildopts.h menuselect.makeopts .lastclean
 	@build_tools/make_version_c > $@.tmp
 	@cmp -s $@.tmp $@ || mv $@.tmp $@
 	@rm -f $@.tmp
@@ -545,7 +545,7 @@ INSTALLDIRS="$(ASTLIBDIR)" "$(ASTMODDIR)" "$(ASTSBINDIR)" "$(ASTCACHEDIR)" "$(AS
 	"$(ASTDATADIR)/firmware/iax" "$(ASTDATADIR)/images" "$(ASTDATADIR)/keys" \
 	"$(ASTDATADIR)/phoneprov" "$(ASTDATADIR)/rest-api" "$(ASTDATADIR)/static-http" \
 	"$(ASTDATADIR)/sounds" "$(ASTDATADIR)/moh" "$(ASTMANDIR)/man8" "$(AGI_DIR)" "$(ASTDBDIR)" \
-	"$(ASTDATADIR)/third-party" "${ASTDATADIR}/keys/stir_shaken"
+	"$(ASTDATADIR)/third-party" "${ASTDATADIR}/keys/stir_shaken" "${ASTDATADIR}/keys/stir_shaken/cache"
 
 installdirs:
 	@for i in $(INSTALLDIRS); do \

README-SERIOUSLY.bestpractices.md

@@ -379,9 +379,8 @@ is set to no.
 
 In Asterisk 12 and later, live_dangerously defaults to no.
 
-
-[voip-security-webinar]: https://www.asterisk.org/security/webinar/
-[blog-sip-security]: http://blogs.digium.com/2009/03/28/sip-security/
+[voip-security-webinar]: https://docs.asterisk.org/Deployment/Important-Security-Considerations/Asterisk-Security-Webinars/
+[blog-sip-security]: https://web.archive.org/web/20171030134647/http://blogs.digium.com/2009/03/28/sip-security/
 [Strong Password Generator]: https://www.strongpasswordgenerator.com
 [Filtering Data]: #filtering-data
 [Proper Device Naming]: #proper-device-naming
@@ -389,4 +388,4 @@ In Asterisk 12 and later, live_dangerously defaults to no.
 [Reducing Pattern Match Typos]: #reducing-pattern-match-typos
 [Manager Class Authorizations]: #manager-class-authorizations
 [Avoid Privilege Escalations]: #avoid-privilege-escalations
-[Important Security Considerations]: https://wiki.asterisk.org/wiki/display/AST/Important+Security+Considerations
+[Important Security Considerations]: https://docs.asterisk.org/Deployment/Important-Security-Considerations/

README.md

@@ -20,7 +20,7 @@ more telephony interfaces than just Internet telephony.  Asterisk also has a
 vast amount of support for traditional PSTN telephony, as well.
 
   For more information on the project itself, please visit the Asterisk
-[home page] and the official [wiki].  In addition you'll find lots
+[home page] and the official [documentation].  In addition you'll find lots
 of information compiled by the Asterisk community at [voip-info.org].
 
   There is a book on Asterisk published by O'Reilly under the Creative Commons
@@ -48,7 +48,7 @@ ANY special hardware, not even a sound card) to install and run Asterisk.
 
 Supported telephony hardware includes:
 * All Analog and Digital Interface cards from [Sangoma]
-* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
+* QuickNet Internet PhoneJack and LineJack
 * any full duplex sound card supported by ALSA, OSS, or PortAudio
 * any ISDN card supported by mISDN on Linux
 * The Xorcom Astribank channel bank
@@ -258,7 +258,7 @@ Asterisk is a trademark of Sangoma Technologies Corporation
 
 [home page]: https://www.asterisk.org
 [support]: https://www.asterisk.org/support
-[wiki]: https://wiki.asterisk.org/
+[documentation]: https://docs.asterisk.org/
 [mailing list]: http://lists.digium.com/mailman/listinfo/asterisk-users
 [chan_dahdi.conf]: configs/samples/chan_dahdi.conf.sample
 [voip-info.org]: http://www.voip-info.org/wiki-Asterisk
@@ -269,4 +269,4 @@ Asterisk is a trademark of Sangoma Technologies Corporation
 [CHANGES]: CHANGES
 [configs]: configs
 [doc]: doc
-[Important Security Considerations]: https://wiki.asterisk.org/wiki/display/AST/Important+Security+Considerations
+[Important Security Considerations]: https://docs.asterisk.org/Deployment/Important-Security-Considerations/

SECURITY.md

@@ -2,7 +2,7 @@
 
 ## Supported Versions
 
-The Asterisk project maintains a [wiki page](https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions) of releases. Each version is listed with its release date, security fix only date, and end of life date. Consult this wiki page to see if the version of Asterisk you are reporting a security vulnerability against is still supported.
+The Asterisk project maintains a [documentation page](https://docs.asterisk.org/About-the-Project/Asterisk-Versions/) of releases. Each version is listed with its release date, security fix only date, and end of life date. Consult this wiki page to see if the version of Asterisk you are reporting a security vulnerability against is still supported.
 
 ## Reporting a Vulnerability
 

apps/app_amd.c

@@ -567,7 +567,7 @@ static int load_config(int reload)
 
 	ast_config_destroy(cfg);
 
-	ast_verb(3, "AMD defaults: initialSilence [%d] greeting [%d] afterGreetingSilence [%d] "
+	ast_verb(5, "AMD defaults: initialSilence [%d] greeting [%d] afterGreetingSilence [%d] "
 		"totalAnalysisTime [%d] minimumWordLength [%d] betweenWordsSilence [%d] maximumNumberOfWords [%d] silenceThreshold [%d] maximumWordLength [%d]\n",
 		dfltInitialSilence, dfltGreeting, dfltAfterGreetingSilence, dfltTotalAnalysisTime,
 		dfltMinimumWordLength, dfltBetweenWordsSilence, dfltMaximumNumberOfWords, dfltSilenceThreshold, dfltMaximumWordLength);

apps/app_audiosocket.c

@@ -61,7 +61,7 @@
 		</syntax>
 		<description>
 			<para>Connects to the given TCP service, then transmits channel audio over that socket.  In turn, audio is received from the socket and sent to the channel.  Only audio frames will be transmitted.</para>
-			<para>Protocol is specified at https://wiki.asterisk.org/wiki/display/AST/AudioSocket</para>
+			<para>Protocol is specified at https://docs.asterisk.org/Configuration/Channel-Drivers/AudioSocket/</para>
 			<para>This application does not automatically answer and should generally be preceeded by an application such as Answer() or Progress().</para>
 		</description>
 	</application>
@@ -180,7 +180,7 @@ static int audiosocket_run(struct ast_channel *chan, const char *id, int svc)
 	chanName = ast_channel_name(chan);
 
 	while (1) {
-
+		ms = -1;
 		targetChan = ast_waitfor_nandfds(&chan, 1, &svc, 1, NULL, &outfd, &ms);
 		if (targetChan) {
 			f = ast_read(chan);

apps/app_authenticate.c

@@ -95,8 +95,17 @@ static const char app[] = "Authenticate";
 				maxdigits have been entered (without requiring the user to press the <literal>#</literal> key).
 				Defaults to 0 - no limit - wait for the user press the <literal>#</literal> key.</para>
 			</parameter>
-			<parameter name="prompt" required="false">
-				<para>Override the agent-pass prompt file.</para>
+			<parameter name="prompt" required="false" argsep="&amp;">
+				<para>Override the &quot;agent-pass&quot; sound file. Can be
+				an ampersand separated list of filenames. If the filename
+				is a relative filename (it does not begin with a slash), it
+				will be searched for in the Asterisk sounds directory. If the
+				filename is able to be parsed as a URL, Asterisk will
+				download the file and then begin playback on it. To include a
+				literal <literal>&amp;</literal> in the URL you can enclose
+				the URL in single quotes.</para>
+				<argument name="prompt" required="true" />
+				<argument name="prompt2" multiple="true" />
 			</parameter>
 		</syntax>
 		<description>

apps/app_chanisavail.c

@@ -117,6 +117,7 @@ static int chanavail_exec(struct ast_channel *chan, const char *data)
 	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");
+	struct ast_format_cap *caps = NULL;
 	AST_DECLARE_APP_ARGS(args,
 		AST_APP_ARG(reqchans);
 		AST_APP_ARG(options);
@@ -126,6 +127,10 @@ static int chanavail_exec(struct ast_channel *chan, const char *data)
 
 	AST_STANDARD_APP_ARGS(args, info);
 
+	ao2_lock(chan);
+	caps = ao2_bump(ast_channel_nativeformats(chan));
+	ao2_unlock(chan);
+
 	if (args.options) {
 		if (strchr(args.options, 'a')) {
 			option_all_avail = 1;
@@ -174,10 +179,11 @@ static int chanavail_exec(struct ast_channel *chan, const char *data)
 			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);
+		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))) {
+			&& (tempchan = ast_request(tech, caps, NULL, chan, number, &status))) {
+
 			ast_str_append(&tmp_availchan, 0, "%s%s",
 				ast_str_strlen(tmp_availchan) ? "&" : "", ast_channel_name(tempchan));
 
@@ -199,8 +205,11 @@ static int chanavail_exec(struct ast_channel *chan, const char *data)
 				break;
 			}
 		}
+
 	}
 
+	ao2_cleanup(caps);
+
 	pbx_builtin_setvar_helper(chan, "AVAILCHAN", ast_str_buffer(tmp_availchan));
 	/* Store the originally used channel too */
 	pbx_builtin_setvar_helper(chan, "AVAILORIGCHAN", ast_str_buffer(tmp_availorig));

apps/app_chanspy.c

@@ -245,6 +245,11 @@
 							</enum>
 						</enumlist>
 					</option>
+					<option name="D">
+						<para>Interleave the audio coming from the channel and the audio coming to the channel in
+						the output audio as a dual channel stream, rather than mix it. Does nothing if 'o'
+						is also set.</para>
+					</option>
 					<option name="e">
 						<argument name="ext" required="true" />
 						<para>Enable <emphasis>enforced</emphasis> mode, so the spying channel can
@@ -393,6 +398,7 @@ enum {
 	OPTION_EXITONHANGUP      = (1 << 18),   /* Hang up when the spied-on channel hangs up. */
 	OPTION_UNIQUEID          = (1 << 19),	/* The chanprefix is a channel uniqueid or fully specified channel name. */
 	OPTION_LONG_QUEUE        = (1 << 20),	/* Allow usage of a long queue to store audio frames. */
+	OPTION_INTERLEAVED       = (1 << 21),	/* Interleave the Read and Write frames in the output frame. */
 };
 
 enum {
@@ -411,6 +417,7 @@ AST_APP_OPTIONS(spy_opts, {
 	AST_APP_OPTION('B', OPTION_BARGE),
 	AST_APP_OPTION_ARG('c', OPTION_DTMF_CYCLE, OPT_ARG_CYCLE),
 	AST_APP_OPTION('d', OPTION_DTMF_SWITCH_MODES),
+	AST_APP_OPTION('D', OPTION_INTERLEAVED),
 	AST_APP_OPTION_ARG('e', OPTION_ENFORCED, OPT_ARG_ENFORCED),
 	AST_APP_OPTION('E', OPTION_EXITONHANGUP),
 	AST_APP_OPTION_ARG('g', OPTION_GROUP, OPT_ARG_GROUP),
@@ -471,6 +478,56 @@ static int spy_generate(struct ast_channel *chan, void *data, int len, int sampl
 	if (ast_test_flag(&csth->flags, OPTION_READONLY)) {
 		/* Option 'o' was set, so don't mix channel audio */
 		f = ast_audiohook_read_frame(&csth->spy_audiohook, samples, AST_AUDIOHOOK_DIRECTION_READ, ast_format_slin);
+	} else if (ast_test_flag(&csth->flags, OPTION_INTERLEAVED)) {
+		/* Option 'D' was set, so mix the spy frame as an interleaved dual channel frame. */
+		int i;
+		struct ast_frame *fr_read = NULL;
+		struct ast_frame *fr_write = NULL;
+		short read_buf[samples];
+		short write_buf[samples];
+		short stereo_buf[samples * 2];
+		struct ast_frame stereo_frame = {
+			.frametype = AST_FRAME_VOICE,
+			.datalen = sizeof(stereo_buf),
+			.samples = samples,
+		};
+
+		f = ast_audiohook_read_frame_all(&csth->spy_audiohook, samples, ast_format_slin, &fr_read, &fr_write);
+		if (f) {
+			ast_frame_free(f, 0);
+			f = NULL;
+		}
+
+		if (fr_read) {
+			memcpy(read_buf, fr_read->data.ptr, sizeof(read_buf));
+		} else {
+			/* silent out the output frame if we can't read the input */
+			memset(read_buf, 0, sizeof(read_buf));
+		}
+
+		if (fr_write) {
+			memcpy(write_buf, fr_write->data.ptr, sizeof(write_buf));
+		} else {
+			memset(write_buf, 0, sizeof(write_buf));
+		}
+
+		for (i = 0; i < samples; i++) {
+			stereo_buf[i*2] = read_buf[i];
+			stereo_buf[i*2+1] = write_buf[i];
+		}
+
+		stereo_frame.data.ptr = stereo_buf;
+		stereo_frame.subclass.format = ast_format_cache_get_slin_by_rate(samples);
+
+		f = ast_frdup(&stereo_frame);
+
+		if (fr_read) {
+			ast_frame_free(fr_read, 0);
+		}
+		if (fr_write) {
+			ast_frame_free(fr_write, 0);
+		}
+
 	} else {
 		f = ast_audiohook_read_frame(&csth->spy_audiohook, samples, AST_AUDIOHOOK_DIRECTION_BOTH, ast_format_slin);
 	}

apps/app_confbridge.c

@@ -3035,7 +3035,7 @@ static int action_playback(struct ast_bridge_channel *bridge_channel, const char
 	char *file_copy = ast_strdupa(playback_file);
 	char *file = NULL;
 
-	while ((file = strsep(&file_copy, "&"))) {
+	while ((file = ast_strsep(&file_copy, '&', AST_STRSEP_STRIP | AST_STRSEP_TRIM))) {
 		if (ast_stream_and_wait(bridge_channel->chan, file, "")) {
 			ast_log(LOG_WARNING, "Failed to playback file %s to channel\n", file);
 			return -1;
@@ -3059,7 +3059,7 @@ static int action_playback_and_continue(struct confbridge_conference *conference
 	char *file_copy = ast_strdupa(playback_file);
 	char *file = NULL;
 
-	while ((file = strsep(&file_copy, "&"))) {
+	while ((file = ast_strsep(&file_copy, '&', AST_STRSEP_STRIP | AST_STRSEP_TRIM))) {
 		if (ast_streamfile(bridge_channel->chan, file, ast_channel_language(bridge_channel->chan))) {
 			ast_log(LOG_WARNING, "Failed to playback file %s to channel\n", file);
 			return -1;

apps/app_dial.c

@@ -88,9 +88,12 @@
 				</argument>
 				<xi:include xpointer="xpointer(/docs/info[@name='Dial_Resource'])" />
 			</parameter>
-			<parameter name="timeout" required="false">
+			<parameter name="timeout" required="false" argsep="^">
 				<para>Specifies the number of seconds we attempt to dial the specified devices.</para>
 				<para>If not specified, this defaults to 136 years.</para>
+				<para>If a second argument is specified, this controls the number of seconds we attempt to dial the specified devices
+				without receiving early media or ringing. If neither progress, ringing, nor voice frames have been received when this
+				timeout expires, the call will be treated as a CHANUNAVAIL. This can be used to skip destinations that may not be responsive.</para>
 			</parameter>
 			<parameter name="options" required="false">
 				<optionlist>
@@ -242,6 +245,10 @@
 					<para>Asterisk will ignore any connected line update requests or any redirecting party
 					update requests it may receive on this dial attempt.</para>
 				</option>
+				<option name="j">
+					<para>Use the initial stream topology of the caller for outgoing channels, even if the caller topology has changed.</para>
+					<para>NOTE: For this option to work, it has to be present in all invocations of Dial that the caller channel goes through.</para>
+				</option>
 				<option name="k">
 					<para>Allow the called party to enable parking of the call by sending
 					the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
@@ -705,6 +712,7 @@ enum {
 #define OPT_RING_WITH_EARLY_MEDIA (1LLU << 43)
 #define OPT_HANGUPCAUSE      (1LLU << 44)
 #define OPT_HEARPULSING      (1LLU << 45)
+#define OPT_TOPOLOGY_PRESERVE (1LLU << 46)
 
 enum {
 	OPT_ARG_ANNOUNCE = 0,
@@ -749,6 +757,7 @@ AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
 	AST_APP_OPTION('H', OPT_CALLER_HANGUP),
 	AST_APP_OPTION('i', OPT_IGNORE_FORWARDING),
 	AST_APP_OPTION('I', OPT_IGNORE_CONNECTEDLINE),
+	AST_APP_OPTION('j', OPT_TOPOLOGY_PRESERVE),
 	AST_APP_OPTION('k', OPT_CALLEE_PARK),
 	AST_APP_OPTION('K', OPT_CALLER_PARK),
 	AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
@@ -808,6 +817,16 @@ struct chanlist {
 
 AST_LIST_HEAD_NOLOCK(dial_head, chanlist);
 
+static void topology_ds_destroy(void *data) {
+	struct ast_stream_topology *top = data;
+	ast_stream_topology_free(top);
+}
+
+static const struct ast_datastore_info topology_ds_info = {
+	.type = "app_dial_topology_preserve",
+	.destroy = topology_ds_destroy,
+};
+
 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode);
 
 static void chanlist_free(struct chanlist *outgoing)
@@ -1181,7 +1200,7 @@ static void set_duration_var(struct ast_channel *chan, const char *var_base, int
 }
 
 static struct ast_channel *wait_for_answer(struct ast_channel *in,
-	struct dial_head *out_chans, int *to, struct ast_flags64 *peerflags,
+	struct dial_head *out_chans, int *to_answer, int *to_progress, struct ast_flags64 *peerflags,
 	char *opt_args[],
 	struct privacy_args *pa,
 	const struct cause_args *num_in, int *result, char *dtmf_progress,
@@ -1194,7 +1213,9 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 {
 	struct cause_args num = *num_in;
 	int prestart = num.busy + num.congestion + num.nochan;
-	int orig = *to;
+	int orig_answer_to = *to_answer;
+	int progress_to_dup = *to_progress;
+	int orig_progress_to = *to_progress;
 	struct ast_channel *peer = NULL;
 	struct chanlist *outgoing = AST_LIST_FIRST(out_chans);
 	/* single is set if only one destination is enabled */
@@ -1222,7 +1243,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 				 * there is no point in continuing.  The bridge
 				 * will just fail if it gets that far.
 				 */
-				*to = -1;
+				*to_answer = -1;
 				strcpy(pa->status, "CONGESTION");
 				ast_channel_publish_dial(in, outgoing->chan, NULL, pa->status);
 				SCOPE_EXIT_RTN_VALUE(NULL, "%s: can't be made compat with %s\n",
@@ -1238,7 +1259,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 
 	is_cc_recall = ast_cc_is_recall(in, &cc_recall_core_id, NULL);
 
-	while ((*to = ast_remaining_ms(start, orig)) && !peer) {
+	while ((*to_answer = ast_remaining_ms(start, orig_answer_to)) && (*to_progress = ast_remaining_ms(start, progress_to_dup)) && !peer) {
 		struct chanlist *o;
 		int pos = 0; /* how many channels do we handle */
 		int numlines = prestart;
@@ -1264,14 +1285,15 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 			} else {
 				ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
 			}
-			*to = 0;
+			*to_answer = 0;
 			if (is_cc_recall) {
 				ast_cc_failed(cc_recall_core_id, "Everyone is busy/congested for the recall. How sad");
 			}
 			SCOPE_EXIT_RTN_VALUE(NULL, "%s: No outgoing channels available\n", ast_channel_name(in));
 		}
-		winner = ast_waitfor_n(watchers, pos, to);
+		winner = ast_waitfor_n(watchers, pos, to_answer);
 		AST_LIST_TRAVERSE(out_chans, o, node) {
+			int res = 0;
 			struct ast_frame *f;
 			struct ast_channel *c = o->chan;
 
@@ -1346,7 +1368,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 					ast_channel_unlock(in);
 				}
 
-				do_forward(o, &num, peerflags, single, caller_entertained, &orig,
+				do_forward(o, &num, peerflags, single, caller_entertained, &orig_answer_to,
 					forced_clid, stored_clid);
 
 				if (o->chan) {
@@ -1483,6 +1505,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 					 * fine for ringing frames to get sent through.
 					 */
 					++num_ringing;
+					*to_progress = -1;
+					progress_to_dup = -1;
 					if (ignore_cc || cc_frame_received || num_ringing == numlines) {
 						ast_verb(3, "%s is ringing\n", ast_channel_name(c));
 						/* Setup early media if appropriate */
@@ -1526,6 +1550,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 							ast_indicate(in, AST_CONTROL_PROGRESS);
 						}
 					}
+					*to_progress = -1;
+					progress_to_dup = -1;
 					if (!sent_progress) {
 						struct timeval now, then;
 						int64_t diff;
@@ -1548,7 +1574,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 								"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),
+							res |= ast_mf_stream(c, (hearpulsing ? NULL : in),
 							(hearpulsing ? in : NULL), mf_progress, 50, 55, 120, 65, 0);
 						}
 						if (!ast_strlen_zero(sf_progress)) {
@@ -1556,7 +1582,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 								"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),
+							res |= ast_sf_stream(c, (hearpulsing ? NULL : in),
 							(hearpulsing ? in : NULL), sf_progress, 0, 0);
 						}
 						if (!ast_strlen_zero(dtmf_progress)) {
@@ -1564,7 +1590,11 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 								"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);
+							res |= ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
+						}
+						if (res) {
+							ast_log(LOG_WARNING, "Called channel %s hung up post-progress before all digits could be sent\n", ast_channel_name(c));
+							goto wait_over;
 						}
 					}
 					ast_channel_publish_dial(in, c, NULL, "PROGRESS");
@@ -1578,7 +1608,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 								"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),
+							res |= ast_mf_stream(c, (hearpulsing ? NULL : in),
 							(hearpulsing ? in : NULL), mf_wink, 50, 55, 120, 65, 0);
 						}
 						if (!ast_strlen_zero(sf_wink)) {
@@ -1586,9 +1616,13 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 								"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),
+							res |= ast_sf_stream(c, (hearpulsing ? NULL : in),
 							(hearpulsing ? in : NULL), sf_wink, 0, 0);
 						}
+						if (res) {
+							ast_log(LOG_WARNING, "Called channel %s hung up post-wink before all digits could be sent\n", ast_channel_name(c));
+							goto wait_over;
+						}
 					}
 					ast_indicate(in, AST_CONTROL_WINK);
 					break;
@@ -1706,6 +1740,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 				if (caller_entertained) {
 					break;
 				}
+				*to_progress = -1;
+				progress_to_dup = -1;
 				/* Fall through */
 			case AST_FRAME_TEXT:
 				if (single && ast_write(in, f)) {
@@ -1734,7 +1770,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 #endif
 			if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP))) {
 				/* Got hung up */
-				*to = -1;
+				*to_answer = -1;
 				strcpy(pa->status, "CANCEL");
 				pa->canceled = 1;
 				publish_dial_end_event(in, out_chans, NULL, pa->status);
@@ -1758,7 +1794,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 					context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
 					if (onedigit_goto(in, context, (char) f->subclass.integer, 1)) {
 						ast_verb(3, "User hit %c to disconnect call.\n", f->subclass.integer);
-						*to = 0;
+						*to_answer = 0;
 						*result = f->subclass.integer;
 						strcpy(pa->status, "CANCEL");
 						pa->canceled = 1;
@@ -1777,7 +1813,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 				if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
 					detect_disconnect(in, f->subclass.integer, &featurecode)) {
 					ast_verb(3, "User requested call disconnect.\n");
-					*to = 0;
+					*to_answer = 0;
 					strcpy(pa->status, "CANCEL");
 					pa->canceled = 1;
 					publish_dial_end_event(in, out_chans, NULL, pa->status);
@@ -1886,9 +1922,15 @@ skip_frame:;
 		}
 	}
 
-	if (!*to || ast_check_hangup(in)) {
-		ast_verb(3, "Nobody picked up in %d ms\n", orig);
+wait_over:
+	if (!*to_answer || ast_check_hangup(in)) {
+		ast_verb(3, "Nobody picked up in %d ms\n", orig_answer_to);
 		publish_dial_end_event(in, out_chans, NULL, "NOANSWER");
+	} else if (!*to_progress) {
+		ast_verb(3, "No early media received in %d ms\n", orig_progress_to);
+		publish_dial_end_event(in, out_chans, NULL, "CHANUNAVAIL");
+		strcpy(pa->status, "CHANUNAVAIL");
+		*to_answer = 0; /* Reset to prevent hangup */
 	}
 
 	if (is_cc_recall) {
@@ -2260,7 +2302,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 	struct chanlist *outgoing;
 	struct chanlist *tmp;
 	struct ast_channel *peer = NULL;
-	int to; /* timeout */
+	int to_answer, to_progress; /* timeouts */
 	struct cause_args num = { chan, 0, 0, 0 };
 	int cause, hanguptreecause = -1;
 
@@ -2316,6 +2358,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 	 */
 	struct ast_party_caller caller;
 	int max_forwards;
+	struct ast_datastore *topology_ds = NULL;
 	SCOPE_ENTER(1, "%s: Data: %s\n", ast_channel_name(chan), data);
 
 	/* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
@@ -2617,7 +2660,21 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 		 */
 		ast_party_connected_line_copy(&tmp->connected, ast_channel_connected(chan));
 
-		topology = ast_stream_topology_clone(ast_channel_get_stream_topology(chan));
+		if (ast_test_flag64(&opts, OPT_TOPOLOGY_PRESERVE)) {
+			topology_ds = ast_channel_datastore_find(chan, &topology_ds_info, NULL);
+
+			if (!topology_ds && (topology_ds = ast_datastore_alloc(&topology_ds_info, NULL))) {
+				topology_ds->data = ast_stream_topology_clone(ast_channel_get_stream_topology(chan));
+				ast_channel_datastore_add(chan, topology_ds);
+			}
+		}
+
+		if (topology_ds) {
+			ao2_ref(topology_ds->data, +1);
+			topology = topology_ds->data;
+		} else {
+			topology = ast_stream_topology_clone(ast_channel_get_stream_topology(chan));
+		}
 
 		ast_channel_unlock(chan);
 
@@ -2856,14 +2913,31 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 	AST_LIST_TRAVERSE_SAFE_END;
 
 	if (ast_strlen_zero(args.timeout)) {
-		to = -1;
+		to_answer = -1;
+		to_progress = -1;
 	} else {
-		to = atoi(args.timeout);
-		if (to > 0)
-			to *= 1000;
-		else {
-			ast_log(LOG_WARNING, "Invalid timeout specified: '%s'. Setting timeout to infinite\n", args.timeout);
-			to = -1;
+		char *anstimeout = strsep(&args.timeout, "^");
+		if (!ast_strlen_zero(anstimeout)) {
+			to_answer = atoi(anstimeout);
+			if (to_answer > 0) {
+				to_answer *= 1000;
+			} else {
+				ast_log(LOG_WARNING, "Invalid answer timeout specified: '%s'. Setting timeout to infinite\n", args.timeout);
+				to_answer = -1;
+			}
+		} else {
+			to_answer = -1;
+		}
+		if (!ast_strlen_zero(args.timeout)) {
+			to_progress = atoi(args.timeout);
+			if (to_progress > 0) {
+				to_progress *= 1000;
+			} else {
+				ast_log(LOG_WARNING, "Invalid progress timeout specified: '%s'. Setting timeout to infinite\n", args.timeout);
+				to_progress = -1;
+			}
+		} else {
+			to_progress = -1;
 		}
 	}
 
@@ -2903,7 +2977,7 @@ 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,
+	peer = wait_for_answer(chan, &out_chans, &to_answer, &to_progress, peerflags, opt_args, &pa, &num, &result,
 		dtmf_progress, mf_progress, mf_wink, sf_progress, sf_wink,
 		(ast_test_flag64(&opts, OPT_HEARPULSING) ? 1 : 0),
 		ignore_cc, &forced_clid, &stored_clid, &config);
@@ -2911,7 +2985,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 	if (!peer) {
 		if (result) {
 			res = result;
-		} else if (to) { /* Musta gotten hung up */
+		} else if (to_answer) { /* Musta gotten hung up */
 			res = -1;
 		} else { /* Nobody answered, next please? */
 			res = 0;

apps/app_directory.c

@@ -39,6 +39,7 @@
 #include "asterisk/say.h"
 #include "asterisk/app.h"
 #include "asterisk/utils.h"
+#include "asterisk/adsi.h"
 
 /*** DOCUMENTATION
 	<application name="Directory" language="en_US">
@@ -111,12 +112,17 @@
 						<para>Skip calling the extension, instead set it in the <variable>DIRECTORY_EXTEN</variable>
 						channel variable.</para>
 					</option>
+					<option name="d">
+						<para>Enable ADSI support for screen phone searching and retrieval
+						of directory results.</para>
+						<para>Additionally, the channel must be ADSI-enabled and you must
+						have an ADSI-compatible (Type III) CPE for this to work.</para>
+					</option>
 				</optionlist>
 				<note><para>Only one of the <replaceable>f</replaceable>, <replaceable>l</replaceable>, or <replaceable>b</replaceable>
 				options may be specified. <emphasis>If more than one is specified</emphasis>, then Directory will act as
 				if <replaceable>b</replaceable> was specified.  The number
 				of characters for the user to type defaults to <literal>3</literal>.</para></note>
-
 			</parameter>
 		</syntax>
 		<description>
@@ -167,6 +173,7 @@ enum {
 	OPT_ALIAS =           (1 << 7),
 	OPT_CONFIG_FILE =     (1 << 8),
 	OPT_SKIP =            (1 << 9),
+	OPT_ADSI =            (1 << 10),
 };
 
 enum {
@@ -200,8 +207,72 @@ AST_APP_OPTIONS(directory_app_options, {
 	AST_APP_OPTION('a', OPT_ALIAS),
 	AST_APP_OPTION_ARG('c', OPT_CONFIG_FILE, OPT_ARG_FILENAME),
 	AST_APP_OPTION('s', OPT_SKIP),
+	AST_APP_OPTION('d', OPT_ADSI), /* (Would've used 'a', but that was taken already) */
 });
 
+static int adsi_search_input(struct ast_channel *chan)
+{
+	unsigned char buf[256];
+	int bytes = 0;
+	unsigned char keys[6];
+
+	memset(keys, 0, sizeof(keys));
+
+	bytes += ast_adsi_display(buf + bytes, ADSI_COMM_PAGE, 3, ADSI_JUST_CENT, 0, " ", "");
+	bytes += ast_adsi_display(buf + bytes, ADSI_COMM_PAGE, 4, ADSI_JUST_CENT, 0, " ", "");
+	bytes += ast_adsi_set_line(buf + bytes, ADSI_COMM_PAGE, 1);
+	bytes += ast_adsi_input_format(buf + bytes, 1, ADSI_DIR_FROM_LEFT, 0, "Query: ***", "");
+	bytes += ast_adsi_input_control(buf + bytes, ADSI_COMM_PAGE, 4, 1, 1, ADSI_JUST_LEFT);
+	bytes += ast_adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 3, "Search", "Search", "#", 1);
+	bytes += ast_adsi_set_keys(buf + bytes, keys);
+	bytes += ast_adsi_voice_mode(buf + bytes, 0);
+
+	ast_debug(3, "Sending ADSI search input screen on %s\n", ast_channel_name(chan));
+
+	return ast_adsi_transmit_message(chan, buf, bytes, ADSI_MSG_DISPLAY);
+}
+
+static int adsi_confirm_match(struct ast_channel *chan, int seq, int total, const char *exten, const char *name, int showexten)
+{
+	unsigned char buf[4096];
+	int alignments[5] = {ADSI_JUST_CENT, ADSI_JUST_CENT, ADSI_JUST_CENT, ADSI_JUST_CENT};
+	char *lines[5] = {NULL, NULL, NULL, NULL, NULL};
+	int x, bytes = 0;
+	unsigned char keys[8];
+	char matchbuf[32];
+
+	snprintf(matchbuf, sizeof(matchbuf), "%d of %d", seq + 1, total); /* Make it 1-indexed for user consumption */
+
+	lines[0] = " "; /* Leave the first line empty so the following lines stand out more */
+	lines[1] = matchbuf;
+	lines[2] = (char*) name;
+
+	if (showexten) {
+		/* If say extension option is set, show it for ADSI as well */
+		lines[3] = (char*) exten;
+	}
+
+	/* Don't use ast_adsi_print here, this way we can send it all at once instead of in 2 transmissions */
+	for (x = 0; lines[x]; x++) {
+		bytes += ast_adsi_display(buf + bytes, ADSI_INFO_PAGE, x + 1, alignments[x], 0, lines[x], "");
+	}
+	bytes += ast_adsi_set_line(buf + bytes, ADSI_INFO_PAGE, 1);
+
+	keys[3] = ADSI_KEY_APPS + 3;
+	keys[4] = ADSI_KEY_APPS + 4;
+	keys[5] = ADSI_KEY_APPS + 5;
+	/* You might think we only need to set the keys up the first time, but nope, we've got to do it each time. */
+	bytes += ast_adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 3, "Dial", "Dial", "1", 0);
+	bytes += ast_adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 4, "Next", "Next", "*", 0);
+	bytes += ast_adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 5, "Exit", "Exit", "#", 0);
+	bytes += ast_adsi_set_keys(buf + bytes, keys);
+	bytes += ast_adsi_voice_mode(buf + bytes, 0);
+
+	ast_debug(3, "Sending ADSI confirmation menu for %s\n", name);
+
+	return ast_adsi_transmit_message(chan, buf, bytes, ADSI_MSG_DISPLAY);
+}
+
 static int compare(const char *text, const char *template)
 {
 	char digit;
@@ -374,6 +445,10 @@ static int select_item_seq(struct ast_channel *chan, struct directory_item **ite
 	for (ptr = items, i = 0; i < count; i++, ptr++) {
 		item = *ptr;
 
+		if (ast_test_flag(flags, OPT_ADSI) && adsi_confirm_match(chan, i, count, item->exten, item->name, ast_test_flag(flags, OPT_SAYEXTENSION))) {
+			return -1;
+		}
+
 		for (loop = 3 ; loop > 0; loop--) {
 			if (!res)
 				res = play_mailbox_owner(chan, item->context, item->exten, item->name, flags);
@@ -933,6 +1008,18 @@ static int directory_exec(struct ast_channel *chan, const char *data)
 	}
 	digits[7] = digit + '0';
 
+	if (ast_test_flag(&flags, OPT_ADSI)) {
+		if (!ast_adsi_available(chan)) {
+			ast_log(LOG_WARNING, "ADSI not available on %s\n", ast_channel_name(chan));
+			ast_clear_flag(&flags, OPT_ADSI);
+		} else {
+			res = ast_adsi_load_session(chan, NULL, 0, 1);
+			if (res < 0) {
+				return res;
+			}
+		}
+	}
+
 	if (ast_channel_state(chan) != AST_STATE_UP) {
 		if (!ast_test_flag(&flags, OPT_NOANSWER)) {
 			/* Otherwise answer unless we're supposed to read while on-hook */
@@ -940,6 +1027,9 @@ static int directory_exec(struct ast_channel *chan, const char *data)
 		}
 	}
 	for (;;) {
+		if (ast_test_flag(&flags, OPT_ADSI) && adsi_search_input(chan)) {
+			return -1;
+		}
 		if (!ast_strlen_zero(dirintro) && !res) {
 			res = ast_stream_and_wait(chan, dirintro, AST_DIGIT_ANY);
 		} else if (!res) {

apps/app_followme.c

@@ -1068,6 +1068,7 @@ static struct ast_channel *findmeexec(struct fm_args *tpargs, struct ast_channel
 		ast_copy_string(num, nm->number, sizeof(num));
 		for (number = num; number; number = rest) {
 			struct ast_channel *outbound;
+			struct ast_format_cap *caps;
 
 			rest = strchr(number, '&');
 			if (rest) {
@@ -1097,8 +1098,15 @@ static struct ast_channel *findmeexec(struct fm_args *tpargs, struct ast_channel
 						? "/n" : "/m");
 			}
 
-			outbound = ast_request("Local", ast_channel_nativeformats(caller), NULL, caller,
-				tmpuser->dialarg, &dg);
+			/* Capture nativeformats reference in case it gets changed */
+			ast_channel_lock(caller);
+			caps = ao2_bump(ast_channel_nativeformats(caller));
+			ast_channel_unlock(caller);
+
+			outbound = ast_request("Local", caps, NULL, caller, tmpuser->dialarg, &dg);
+
+			ao2_cleanup(caps);
+
 			if (!outbound) {
 				ast_log(LOG_WARNING, "Unable to allocate a channel for Local/%s cause: %s\n",
 					tmpuser->dialarg, ast_cause2str(dg));

apps/app_if.c

@@ -196,6 +196,7 @@ static int find_matching_endif(struct ast_channel *chan, const char *otherapp)
 		if (!ast_rdlock_context(c)) {
 			if (!strcmp(ast_get_context_name(c), ast_channel_context(chan))) {
 				/* This is the matching context we want */
+
 				int cur_priority = ast_channel_priority(chan) + 1, level = 1;
 
 				for (e = find_matching_priority(c, ast_channel_exten(chan), cur_priority,
@@ -203,6 +204,7 @@ static int find_matching_endif(struct ast_channel *chan, const char *otherapp)
 					e;
 					e = find_matching_priority(c, ast_channel_exten(chan), ++cur_priority,
 						S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL))) {
+
 					if (!strcasecmp(ast_get_extension_app(e), "IF")) {
 						level++;
 					} else if (!strcasecmp(ast_get_extension_app(e), "ENDIF")) {
@@ -283,7 +285,12 @@ static int if_helper(struct ast_channel *chan, const char *data, int end)
 		pbx_builtin_setvar_helper(chan, my_name, NULL);
 		snprintf(end_varname,sizeof(end_varname),"END_%s",varname);
 		ast_channel_lock(chan);
-		endifpri = find_matching_endif(chan, NULL);
+		/* For EndIf, simply go to the next priority.
+		 * We do not add 1 to ast_channel_priority because the dialplan will
+		 * auto-increment the priority when we return, so just keep the priority as is.
+		 * For ExitIf or false If() condition, we need to find the end of the current
+		 * If branch (at same indentation) and branch there. */
+		endifpri = end == 2 ? ast_channel_priority(chan) : find_matching_endif(chan, NULL);
 		if ((goto_str = pbx_builtin_getvar_helper(chan, end_varname))) {
 			ast_parseable_goto(chan, goto_str);
 			pbx_builtin_setvar_helper(chan, end_varname, NULL);

apps/app_playback.c

@@ -48,6 +48,13 @@
 		</synopsis>
 		<syntax>
 			<parameter name="filenames" required="true" argsep="&amp;">
+				<para>Ampersand separated list of filenames. If the filename
+				is a relative filename (it does not begin with a slash), it
+				will be searched for in the Asterisk sounds directory. If the
+				filename is able to be parsed as a URL, Asterisk will
+				download the file and then begin playback on it. To include a
+				literal <literal>&amp;</literal> in the URL you can enclose
+				the URL in single quotes.</para>
 				<argument name="filename" required="true" />
 				<argument name="filename2" multiple="true" />
 			</parameter>
@@ -492,7 +499,7 @@ static int playback_exec(struct ast_channel *chan, const char *data)
 		char *front;
 
 		ast_stopstream(chan);
-		while (!res && (front = strsep(&back, "&"))) {
+		while (!res && (front = ast_strsep(&back, '&', AST_STRSEP_STRIP | AST_STRSEP_TRIM))) {
 			if (option_say)
 				res = say_full(chan, front, "", ast_channel_language(chan), NULL, -1, -1);
 			else if (option_mix){

apps/app_queue.c

@@ -230,11 +230,18 @@
 				<para><replaceable>URL</replaceable> will be sent to the called party if the channel supports it.</para>
 			</parameter>
 			<parameter name="announceoverride" argsep="&amp;">
-				<argument name="filename" required="true">
-					<para>Announcement file(s) to play to agent before bridging call, overriding the announcement(s)
-					configured in <filename>queues.conf</filename>, if any.</para>
-				</argument>
-				<argument name="filename2" multiple="true" />
+				<para>Announcement file(s) to play to agent before bridging
+				call, overriding the announcement(s) configured in
+				<filename>queues.conf</filename>, if any.</para>
+				<para>Ampersand separated list of filenames. If the filename
+				is a relative filename (it does not begin with a slash), it
+				will be searched for in the Asterisk sounds directory. If the
+				filename is able to be parsed as a URL, Asterisk will
+				download the file and then begin playback on it. To include a
+				literal <literal>&amp;</literal> in the URL you can enclose
+				the URL in single quotes.</para>
+				<argument name="announceoverride" required="true" />
+				<argument name="announceoverride2" multiple="true" />
 			</parameter>
 			<parameter name="timeout">
 				<para>Will cause the queue to fail out after a specified number of
@@ -1845,6 +1852,7 @@ struct call_queue {
 	int announcepositionlimit;          /*!< How many positions we announce? */
 	int announcefrequency;              /*!< How often to announce their position */
 	int minannouncefrequency;           /*!< The minimum number of seconds between position announcements (def. 15) */
+	int periodicannouncestartdelay;     /*!< How long into the queue should the periodic accouncement start */
 	int periodicannouncefrequency;      /*!< How often to play periodic announcement */
 	int numperiodicannounce;            /*!< The number of periodic announcements configured */
 	int randomperiodicannounce;         /*!< Are periodic announcments randomly chosen */
@@ -2975,6 +2983,7 @@ static void init_queue(struct call_queue *q)
 	q->weight = 0;
 	q->timeoutrestart = 0;
 	q->periodicannouncefrequency = 0;
+	q->periodicannouncestartdelay = -1;
 	q->randomperiodicannounce = 0;
 	q->numperiodicannounce = 0;
 	q->relativeperiodicannounce = 0;
@@ -3431,6 +3440,8 @@ static void queue_set_param(struct call_queue *q, const char *param, const char
 			ast_str_set(&q->sound_periodicannounce[0], 0, "%s", val);
 			q->numperiodicannounce = 1;
 		}
+	} else if (!strcasecmp(param, "periodic-announce-startdelay")) {
+		q->periodicannouncestartdelay = atoi(val);
 	} else if (!strcasecmp(param, "periodic-announce-frequency")) {
 		q->periodicannouncefrequency = atoi(val);
 	} else if (!strcasecmp(param, "relative-periodic-announce")) {
@@ -7191,7 +7202,7 @@ static int try_calling(struct queue_ent *qe, struct ast_flags opts, char **opt_a
 				if (!res2 && announce) {
 					char *front;
 					char *announcefiles = ast_strdupa(announce);
-					while ((front = strsep(&announcefiles, "&"))) {
+					while ((front = ast_strsep(&announcefiles, '&', AST_STRSEP_STRIP | AST_STRSEP_TRIM))) {
 						if (play_file(peer, front) < 0) {
 							ast_log(LOG_ERROR, "play_file failed for '%s' on %s\n", front, ast_channel_name(peer));
 						}
@@ -7794,6 +7805,9 @@ static void set_queue_member_pause(struct call_queue *q, struct member *mem, con
 	if (paused && !ast_strlen_zero(reason)) {
 		ast_copy_string(mem->reason_paused, reason, sizeof(mem->reason_paused));
 	} else {
+		/* We end up filling this in again later (temporarily) but we need it
+		 * empty for now so that the intervening code - specifically
+		 * dump_queue_members() - has the correct view of things. */
 		mem->reason_paused[0] = '\0';
 	}
 
@@ -7812,10 +7826,22 @@ static void set_queue_member_pause(struct call_queue *q, struct member *mem, con
 			"Queue:%s_avail", q->name);
 	}
 
-	ast_queue_log(q->name, "NONE", mem->membername, (paused ? "PAUSE" : "UNPAUSE"),
-		"%s", S_OR(reason, ""));
+	if (!paused && !ast_strlen_zero(reason)) {
+		/* Because we've been unpaused with a 'reason' we need to ensure that
+		 * that reason is emitted when the subsequent PauseQueueMember event
+		 * is raised. So temporarily set it on the member and clear it out
+		 * again right after. */
+		ast_copy_string(mem->reason_paused, reason, sizeof(mem->reason_paused));
+	}
+
+	ast_queue_log(q->name, "NONE", mem->membername, paused ? "PAUSE" : "UNPAUSE",
+		"%s", mem->reason_paused);
 
 	publish_queue_member_pause(q, mem);
+
+	if (!paused) {
+		mem->reason_paused[0] = '\0';
+	}
 }
 
 static int set_member_paused(const char *queuename, const char *interface, const char *reason, int paused)
@@ -8652,6 +8678,11 @@ static int queue_exec(struct ast_channel *chan, const char *data)
 	}
 	ast_assert(qe.parent != NULL);
 
+	if (qe.parent->periodicannouncestartdelay >= 0) {
+		qe.last_periodic_announce_time += qe.parent->periodicannouncestartdelay;
+		qe.last_periodic_announce_time -= qe.parent->periodicannouncefrequency;
+	}
+
 	ast_queue_log(args.queuename, ast_channel_uniqueid(chan), "NONE", "ENTERQUEUE", "%s|%s|%d",
 		S_OR(args.url, ""),
 		S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, ""),
@@ -10957,7 +10988,7 @@ static char *handle_queue_add_member(struct ast_cli_entry *e, int cmd, struct as
 	case CLI_INIT:
 		e->command = "queue add member";
 		e->usage =
-			"Usage: queue add member <dial string> to <queue> [[[penalty <penalty>] as <membername>] state_interface <interface>]\n"
+			"Usage: queue add member <dial string> to <queue> [penalty <penalty> [as <membername> [state_interface <interface>]]]\n"
 			"       Add a dial string (Such as a channel,e.g. SIP/6001) to a queue with optionally:  a penalty, membername and a state_interface\n";
 		return NULL;
 	case CLI_GENERATE:

apps/app_read.c

@@ -49,9 +49,15 @@
 				name.</para>
 			</parameter>
 			<parameter name="filenames" argsep="&amp;">
-				<argument name="filename" required="true">
-					<para>file(s) to play before reading digits or tone with option i</para>
-				</argument>
+				<para>Ampersand separated list of filenames to play before
+				reading digits or tone with option <literal>i</literal>. If
+				the filename is a relative filename (it does not begin with a
+				slash), it will be searched for in the Asterisk sounds
+				directory. If the filename is able to be parsed as a URL,
+				Asterisk will download the file and then begin playback on
+				it. To include a literal <literal>&amp;</literal> in the URL
+				you can enclose	the URL in single quotes.</para>
+				<argument name="filename" required="true" />
 				<argument name="filename2" multiple="true" />
 			</parameter>
 			<parameter name="maxdigits">

apps/app_skel.c

@@ -16,7 +16,7 @@
  * at the top of the source tree.
  *
  * Please follow coding guidelines
- * https://wiki.asterisk.org/wiki/display/AST/Coding+Guidelines
+ * https://docs.asterisk.org/Development/Policies-and-Procedures/Coding-Guidelines/
  */
 
 /*! \file
@@ -371,7 +371,8 @@ static void play_files_helper(struct ast_channel *chan, const char *prompts)
 	char *prompt, *rest = ast_strdupa(prompts);
 
 	ast_stopstream(chan);
-	while ((prompt = strsep(&rest, "&")) && !ast_stream_and_wait(chan, prompt, "")) {
+	while ((prompt = ast_strsep(&rest, '&', AST_STRSEP_STRIP | AST_STRSEP_TRIM))
+		&& !ast_stream_and_wait(chan, prompt, "")) {
 		ast_stopstream(chan);
 	}
 }

apps/app_speech_utils.c

@@ -85,7 +85,17 @@
 			Play a sound file and wait for speech to be recognized.
 		</synopsis>
 		<syntax>
-			<parameter name="sound_file" required="true" />
+			<parameter name="sound_file" required="true" argsep="&amp;">
+				<para>Ampersand separated list of filenames. If the filename
+				is a relative filename (it does not begin with a slash), it
+				will be searched for in the Asterisk sounds directory. If the
+				filename is able to be parsed as a URL, Asterisk will
+				download the file and then begin playback on it. To include a
+				literal <literal>&amp;</literal> in the URL you can enclose
+				the URL in single quotes.</para>
+				<argument name="sound_file" required="true" />
+				<argument name="sound_file2" multiple="true" />
+			</parameter>
 			<parameter name="timeout">
 				<para>Timeout integer in seconds. Note the timeout will only start
 				once the sound file has stopped playing.</para>
@@ -95,6 +105,9 @@
 					<option name="n">
 						<para>Don't answer the channel if it has not already been answered.</para>
 					</option>
+					<option name="p">
+						<para>Return partial results when backend is terminated by timeout.</para>
+					</option>
 				</optionlist>
 			</parameter>
 		</syntax>
@@ -690,10 +703,12 @@ static int speech_streamfile(struct ast_channel *chan, const char *filename, con
 
 enum {
 	SB_OPT_NOANSWER = (1 << 0),
+	SB_OPT_PARTIALRESULTS = (1 << 1),
 };
 
 AST_APP_OPTIONS(speech_background_options, BEGIN_OPTIONS
 	AST_APP_OPTION('n', SB_OPT_NOANSWER),
+	AST_APP_OPTION('p', SB_OPT_PARTIALRESULTS),
 END_OPTIONS );
 
 /*! \brief SpeechBackground(Sound File,Timeout) Dialplan Application */
@@ -776,7 +791,10 @@ static int speech_background(struct ast_channel *chan, const char *data)
 	/* Okay it's streaming so go into a loop grabbing frames! */
 	while (done == 0) {
 		/* If the filename is null and stream is not running, start up a new sound file */
-		if (!quieted && (ast_channel_streamid(chan) == -1 && ast_channel_timingfunc(chan) == NULL) && (filename = strsep(&filename_tmp, "&"))) {
+		if (!quieted
+			&& ast_channel_streamid(chan) == -1
+			&& ast_channel_timingfunc(chan) == NULL
+			&& (filename = ast_strsep(&filename_tmp, '&', AST_STRSEP_STRIP | AST_STRSEP_TRIM))) {
 			/* Discard old stream information */
 			ast_stopstream(chan);
 			/* Start new stream */
@@ -920,7 +938,10 @@ static int speech_background(struct ast_channel *chan, const char *data)
 		}
 	}
 
-	if (!ast_strlen_zero(dtmf)) {
+	if (ast_strlen_zero(dtmf) && speech->state == AST_SPEECH_STATE_READY && ast_test_flag(&options, SB_OPT_PARTIALRESULTS)) {
+		/* Copy to speech structure the results, even partial ones, if desired and available */
+		speech->results = ast_speech_results_get(speech);
+	} else if (!ast_strlen_zero(dtmf)) {
 		/* We sort of make a results entry */
 		speech->results = ast_calloc(1, sizeof(*speech->results));
 		if (speech->results != NULL) {

apps/app_stack.c

@@ -375,26 +375,6 @@ static int pop_exec(struct ast_channel *chan, const char *data)
 	return res;
 }
 
-static int frames_left(struct ast_channel *chan)
-{
-	struct ast_datastore *stack_store;
-	struct gosub_stack_list *oldlist;
-	int exists;
-
-	ast_channel_lock(chan);
-	stack_store = ast_channel_datastore_find(chan, &stack_info, NULL);
-	if (!stack_store) {
-		ast_channel_unlock(chan);
-		return -1;
-	}
-	oldlist = stack_store->data;
-	AST_LIST_LOCK(oldlist);
-	exists = oldlist->first ? 1 : 0;
-	AST_LIST_UNLOCK(oldlist);
-	ast_channel_unlock(chan);
-	return exists;
-}
-
 static int return_exec(struct ast_channel *chan, const char *data)
 {
 	struct ast_datastore *stack_store;
@@ -402,7 +382,6 @@ static int return_exec(struct ast_channel *chan, const char *data)
 	struct gosub_stack_list *oldlist;
 	const char *retval = data;
 	int res = 0;
-	int lastframe;
 
 	ast_channel_lock(chan);
 	if (!(stack_store = ast_channel_datastore_find(chan, &stack_info, NULL))) {
@@ -414,7 +393,6 @@ static int return_exec(struct ast_channel *chan, const char *data)
 	oldlist = stack_store->data;
 	AST_LIST_LOCK(oldlist);
 	oldframe = AST_LIST_REMOVE_HEAD(oldlist, entries);
-	lastframe = oldlist->first ? 0 : 1;
 	AST_LIST_UNLOCK(oldlist);
 
 	if (!oldframe) {
@@ -432,19 +410,12 @@ static int return_exec(struct ast_channel *chan, const char *data)
 	 * what was there before.  Channels that do not have a PBX may
 	 * not have the context or exten set.
 	 */
-	if (ast_channel_pbx(chan) || !lastframe) {
-		/* If there's no PBX, the "old location" is simply
-		 * the configured context for the device, such as
-		 * for pre-dial handlers, and restoring this location
-		 * is nonsensical. So if no PBX and there are no further
-		 * frames, leave the location as it is. */
-		ast_channel_context_set(chan, oldframe->context);
-		ast_channel_exten_set(chan, oldframe->extension);
-		if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_IN_AUTOLOOP)) {
-			--oldframe->priority;
-		}
-		ast_channel_priority_set(chan, oldframe->priority);
+	ast_channel_context_set(chan, oldframe->context);
+	ast_channel_exten_set(chan, oldframe->extension);
+	if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_IN_AUTOLOOP)) {
+		--oldframe->priority;
 	}
+	ast_channel_priority_set(chan, oldframe->priority);
 	ast_set2_flag(ast_channel_flags(chan), oldframe->in_subroutine, AST_FLAG_SUBROUTINE_EXEC);
 
 	gosub_release_frame(chan, oldframe);
@@ -1095,13 +1066,10 @@ static int gosub_run(struct ast_channel *chan, const char *sub_args, int ignore_
 				ast_channel_priority(chan), ast_channel_name(chan));
 		}
 
-		/* Did the routine return?
-		 * For things like predial where there's no PBX on the channel yet,
-		 * the last return leaves the location alone so we can print it out correctly here.
-		 * So to ensure we finished properly, make sure there are no frames left in that case. */
-		if ((!ast_channel_pbx(chan) && !frames_left(chan)) || (ast_channel_priority(chan) == saved_priority
+		/* Did the routine return? */
+		if (ast_channel_priority(chan) == saved_priority
 			&& !strcmp(ast_channel_context(chan), saved_context)
-			&& !strcmp(ast_channel_exten(chan), saved_exten))) {
+			&& !strcmp(ast_channel_exten(chan), saved_exten)) {
 			ast_verb(3, "%s Internal %s(%s) complete GOSUB_RETVAL=%s\n",
 				ast_channel_name(chan), app_gosub, sub_args,
 				S_OR(pbx_builtin_getvar_helper(chan, "GOSUB_RETVAL"), ""));

apps/app_voicemail.c

@@ -27,7 +27,7 @@
  *
  * \par See also
  * \arg \ref voicemail.conf "Config_voicemail"
- * \note For information about voicemail IMAP storage, https://wiki.asterisk.org/wiki/display/AST/IMAP+Voicemail+Storage
+ * \note For information about voicemail IMAP storage, https://docs.asterisk.org/Configuration/Applications/Voicemail/IMAP-Voicemail-Storage/
  * \ingroup applications
  * \todo This module requires res_adsi to load. This needs to be optional
  * during compilation.
@@ -546,6 +546,22 @@
 			as the same as the from.</para>
 		</description>
 	</manager>
+	<managerEvent language="en_US" name="VoicemailPasswordChange">
+		<managerEventInstance class="EVENT_FLAG_USER">
+			<synopsis>Raised in response to a mailbox password change.</synopsis>
+			<syntax>
+				<parameter name="Context">
+					<para>Mailbox context.</para>
+				</parameter>
+				<parameter name="Mailbox">
+					<para>Mailbox name.</para>
+				</parameter>
+				<parameter name="NewPassword">
+					<para>New password for mailbox.</para>
+				</parameter>
+			</syntax>
+		</managerEventInstance>
+	</managerEvent>
  ***/
 
 #ifdef IMAP_STORAGE
@@ -672,6 +688,7 @@ static AST_LIST_HEAD_STATIC(vmstates, vmstate);
 #define VM_MESSAGEWRAP   (1 << 17)  /*!< Wrap around from the last message to the first, and vice-versa */
 #define VM_FWDURGAUTO    (1 << 18)  /*!< Autoset of Urgent flag on forwarded Urgent messages set globally */
 #define VM_EMAIL_EXT_RECS (1 << 19)  /*!< Send voicemail emails when an external recording is added to a mailbox */
+#define VM_MARK_URGENT   (1 << 20)	/*!< After recording, permit the caller to mark the message as urgent */
 #define ERROR_LOCK_PATH  -100
 #define ERROR_MAX_MSGS   -101
 #define OPERATOR_EXIT     300
@@ -751,6 +768,18 @@ static const char * const mailbox_folders[] = {
 	"Urgent",
 };
 
+/*!
+ * \brief Reload voicemail.conf
+ * \param reload Whether this is a reload as opposed to module load
+ * \param force Forcefully reload the config, even it has not changed
+ * \retval 0 on success, nonzero on failure
+ */
+static int load_config_force(int reload, int force);
+
+/*! \brief Forcibly reload voicemail.conf, even if it has not changed.
+ * This is necessary after running unit tests. */
+#define force_reload_config() load_config_force(1, 1)
+
 static int load_config(int reload);
 #ifdef TEST_FRAMEWORK
 static int load_config_from_memory(int reload, struct ast_config *cfg, struct ast_config *ucfg);
@@ -1410,6 +1439,8 @@ static void apply_option(struct ast_vm_user *vmu, const char *var, const char *v
 		ast_set2_flag(vmu, ast_true(value), VM_SVMAIL);
 	} else if (!strcasecmp(var, "review")){
 		ast_set2_flag(vmu, ast_true(value), VM_REVIEW);
+	} else if (!strcasecmp(var, "leaveurgent")){
+		ast_set2_flag(vmu, ast_true(value), VM_MARK_URGENT);
 	} else if (!strcasecmp(var, "tempgreetwarn")){
 		ast_set2_flag(vmu, ast_true(value), VM_TEMPGREETWARN);
 	} else if (!strcasecmp(var, "messagewrap")){
@@ -1849,6 +1880,16 @@ static int reset_user_pw(const char *context, const char *mailbox, const char *n
 		res = 0;
 	}
 	AST_LIST_UNLOCK(&users);
+	if (!res) {
+		struct ast_json *json_object;
+
+		json_object = ast_json_pack("{s: s, s: s, s: s}",
+			"Context", S_OR(context, "default"),
+			"Mailbox", mailbox,
+			"NewPassword", newpass);
+		ast_manager_publish_event("VoicemailPasswordChange", EVENT_FLAG_SYSTEM | EVENT_FLAG_USER, json_object);
+		ast_json_unref(json_object);
+	}
 	return res;
 }
 
@@ -1892,7 +1933,7 @@ static void vm_change_password(struct ast_vm_user *vmu, const char *newpassword)
 			ast_copy_string(vmu->password, newpassword, sizeof(vmu->password));
 			break;
 		} else {
-			ast_verb(4, "Writing voicemail password to file %s failed, falling back to config file\n", secretfn);
+			ast_log(LOG_WARNING, "Writing voicemail password to file %s failed, falling back to config file\n", secretfn);
 		}
 		/* Fall-through */
 	case OPT_PWLOC_VOICEMAILCONF:
@@ -4086,16 +4127,14 @@ bail:
 
 /*!
  * \brief Determines the highest message number in use for a given user and mailbox folder.
- * \param vmu
  * \param dir the folder the mailbox folder to look for messages. Used to construct the SQL where clause.
  *
  * This method is used when mailboxes are stored in an ODBC back end.
  * Typical use to set the msgnum would be to take the value returned from this method and add one to it.
  *
  * \return the value of zero or greater to indicate the last message index in use, -1 to indicate none.
-
  */
-static int last_message_index(struct ast_vm_user *vmu, char *dir)
+static int last_message_index(char *dir)
 {
 	int x = -1;
 	int res;
@@ -4380,15 +4419,15 @@ static SQLHSTMT insert_data_cb(struct odbc_obj *obj, void *vdata)
 	SQLBindParameter(stmt, 2, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->msgnums), 0, (void *) data->msgnums, 0, NULL);
 	SQLBindParameter(stmt, 3, SQL_PARAM_INPUT, SQL_C_BINARY, SQL_LONGVARBINARY, data->datalen, 0, (void *) data->data, data->datalen, &data->indlen);
 	SQLBindParameter(stmt, 4, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->context), 0, (void *) data->context, 0, NULL);
-	SQLBindParameter(stmt, 6, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->callerid), 0, (void *) data->callerid, 0, NULL);
-	SQLBindParameter(stmt, 7, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->origtime), 0, (void *) data->origtime, 0, NULL);
-	SQLBindParameter(stmt, 8, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->duration), 0, (void *) data->duration, 0, NULL);
-	SQLBindParameter(stmt, 9, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->mailboxuser), 0, (void *) data->mailboxuser, 0, NULL);
-	SQLBindParameter(stmt, 10, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->mailboxcontext), 0, (void *) data->mailboxcontext, 0, NULL);
-	SQLBindParameter(stmt, 11, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->flag), 0, (void *) data->flag, 0, NULL);
-	SQLBindParameter(stmt, 12, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->msg_id), 0, (void *) data->msg_id, 0, NULL);
+	SQLBindParameter(stmt, 5, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->callerid), 0, (void *) data->callerid, 0, NULL);
+	SQLBindParameter(stmt, 6, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->origtime), 0, (void *) data->origtime, 0, NULL);
+	SQLBindParameter(stmt, 7, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->duration), 0, (void *) data->duration, 0, NULL);
+	SQLBindParameter(stmt, 8, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->mailboxuser), 0, (void *) data->mailboxuser, 0, NULL);
+	SQLBindParameter(stmt, 9, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->mailboxcontext), 0, (void *) data->mailboxcontext, 0, NULL);
+	SQLBindParameter(stmt, 10, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->flag), 0, (void *) data->flag, 0, NULL);
+	SQLBindParameter(stmt, 11, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->msg_id), 0, (void *) data->msg_id, 0, NULL);
 	if (!ast_strlen_zero(data->category)) {
-		SQLBindParameter(stmt, 13, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->category), 0, (void *) data->category, 0, NULL);
+		SQLBindParameter(stmt, 12, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(data->category), 0, (void *) data->category, 0, NULL);
 	}
 	res = ast_odbc_execute_sql(obj, stmt, data->sql);
 	if (!SQL_SUCCEEDED(res)) {
@@ -4658,7 +4697,6 @@ static void rename_file(char *sfn, char *dfn)
 
 /*!
  * \brief Determines the highest message number in use for a given user and mailbox folder.
- * \param vmu
  * \param dir the folder the mailbox folder to look for messages. Used to construct the SQL where clause.
  *
  * This method is used when mailboxes are stored on the filesystem. (not ODBC and not IMAP).
@@ -4667,7 +4705,7 @@ static void rename_file(char *sfn, char *dfn)
  * \note Should always be called with a lock already set on dir.
  * \return the value of zero or greaterto indicate the last message index in use, -1 to indicate none.
  */
-static int last_message_index(struct ast_vm_user *vmu, char *dir)
+static int last_message_index(char *dir)
 {
 	int x;
 	unsigned char map[MAXMSGLIMIT] = "";
@@ -4694,12 +4732,8 @@ static int last_message_index(struct ast_vm_user *vmu, char *dir)
 	}
 	closedir(msgdir);
 
-	for (x = 0; x < vmu->maxmsg; x++) {
-		if (map[x] == 1) {
-			stopcount--;
-		} else if (map[x] == 0 && !stopcount) {
-			break;
-		}
+	for (x = 0; x < MAXMSGLIMIT && stopcount; x++) {
+		stopcount -= map[x];
 	}
 
 	return x - 1;
@@ -5972,7 +6006,7 @@ static int copy_message(struct ast_channel *chan, struct ast_vm_user *vmu, int i
 	if (vm_lock_path(todir))
 		return ERROR_LOCK_PATH;
 
-	recipmsgnum = last_message_index(recip, todir) + 1;
+	recipmsgnum = last_message_index(todir) + 1;
 	if (recipmsgnum < recip->maxmsg - (imbox ? 0 : inprocess_count(vmu->mailbox, vmu->context, 0))) {
 		make_file(topath, sizeof(topath), todir, recipmsgnum);
 #ifndef ODBC_STORAGE
@@ -6467,7 +6501,7 @@ static int msg_create_from_file(struct ast_vm_recording_data *recdata)
 		return -1;
 	}
 
-	msgnum = last_message_index(recipient, dir) + 1;
+	msgnum = last_message_index(dir) + 1;
 #endif
 
 	/* Lock the directory receiving the voicemail since we want it to still exist when we attempt to copy the voicemail.
@@ -7035,7 +7069,7 @@ static int leave_voicemail(struct ast_channel *chan, char *ext, struct leave_vm_
 					}
 				} else {
 #ifndef IMAP_STORAGE
-					msgnum = last_message_index(vmu, dir) + 1;
+					msgnum = last_message_index(dir) + 1;
 #endif
 					make_file(fn, sizeof(fn), dir, msgnum);
 
@@ -7157,7 +7191,7 @@ static int resequence_mailbox(struct ast_vm_user *vmu, char *dir, int stopcount)
 		return ERROR_LOCK_PATH;
 	}
 
-	for (x = 0, dest = 0; dest != stopcount && x < vmu->maxmsg + 10; x++) {
+	for (x = 0, dest = 0; dest != stopcount && x < MAXMSGLIMIT; x++) {
 		make_file(sfn, sizeof(sfn), dir, x);
 		if (EXISTS(dir, x, sfn, NULL)) {
 
@@ -7246,7 +7280,7 @@ static int save_to_folder(struct ast_vm_user *vmu, struct vm_state *vms, int msg
 	if (vm_lock_path(ddir))
 		return ERROR_LOCK_PATH;
 
-	x = last_message_index(vmu, ddir) + 1;
+	x = last_message_index(ddir) + 1;
 
 	if (box == 10 && x >= vmu->maxdeletedmsg) { /* "Deleted" folder*/
 		x--;
@@ -7422,8 +7456,11 @@ static void adsi_begin(struct ast_channel *chan, int *useadsi)
 	if (!ast_adsi_available(chan))
 		return;
 	x = ast_adsi_load_session(chan, adsifdn, adsiver, 1);
-	if (x < 0)
+	if (x < 0) {
+		*useadsi = 0;
+		ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE);
 		return;
+	}
 	if (!x) {
 		if (adsi_load_vmail(chan, useadsi)) {
 			ast_log(AST_LOG_WARNING, "Unable to upload voicemail scripts\n");
@@ -9103,8 +9140,8 @@ static int open_mailbox(struct vm_state *vms, struct ast_vm_user *vmu, int box)
 		return ERROR_LOCK_PATH;
 	}
 
-	/* for local storage, checks directory for messages up to maxmsg limit */
-	last_msg = last_message_index(vmu, vms->curdir);
+	/* for local storage, checks directory for messages up to MAXMSGLIMIT */
+	last_msg = last_message_index(vms->curdir);
 	ast_unlock_path(vms->curdir);
 
 	if (last_msg < -1) {
@@ -9140,7 +9177,7 @@ static int close_mailbox(struct vm_state *vms, struct ast_vm_user *vmu)
 	}
 
 	/* update count as message may have arrived while we've got mailbox open */
-	last_msg_idx = last_message_index(vmu, vms->curdir);
+	last_msg_idx = last_message_index(vms->curdir);
 	if (last_msg_idx != vms->lastmsg) {
 		ast_log(AST_LOG_NOTICE, "%d messages received after mailbox opened.\n", last_msg_idx - vms->lastmsg);
 	}
@@ -11550,6 +11587,7 @@ static int show_mailbox_snapshot(struct ast_cli_args *a)
 	const char *context = a->argv[4];
 	struct ast_vm_mailbox_snapshot *mailbox_snapshot;
 	struct ast_vm_msg_snapshot *msg;
+	int i;
 
 	/* Take a snapshot of the mailbox and walk through each folder's contents */
 	mailbox_snapshot = ast_vm_mailbox_snapshot_create(mailbox, context, NULL, 0, AST_VM_SNAPSHOT_SORT_BY_ID, 0);
@@ -11560,7 +11598,7 @@ static int show_mailbox_snapshot(struct ast_cli_args *a)
 
 	ast_cli(a->fd, VM_STRING_HEADER_FORMAT, "Folder", "Caller ID", "Date", "Duration", "Flag", "ID");
 
-	for (int i = 0; i < mailbox_snapshot->folders; i++) {
+	for (i = 0; i < mailbox_snapshot->folders; i++) {
 		AST_LIST_TRAVERSE(&((mailbox_snapshot)->snapshots[i]), msg, msg) {
 			ast_cli(a->fd, VM_STRING_HEADER_FORMAT, msg->folder_name, msg->callerid, msg->origdate, msg->duration,
 					msg->flag, msg->msg_id);
@@ -11777,9 +11815,10 @@ static char *complete_voicemail_move_message(struct ast_cli_args *a, int maxpos)
 		}
 		AST_LIST_UNLOCK(&users);
 	} else if (pos == 4 || pos == 8 || (pos == 6 && maxpos == 6) ) {
+		int i;
 		/* Walk through the standard folders */
 		wordlen = strlen(word);
-		for (int i = 0; i < ARRAY_LEN(mailbox_folders); i++) {
+		for (i = 0; i < ARRAY_LEN(mailbox_folders); i++) {
 			if (folder && !strncasecmp(word, mailbox_folders[i], wordlen) && ++which > state) {
 				return ast_strdup(mailbox_folders[i]);
 			}
@@ -11789,7 +11828,6 @@ static char *complete_voicemail_move_message(struct ast_cli_args *a, int maxpos)
 		/* find messages in the folder */
 		struct ast_vm_mailbox_snapshot *mailbox_snapshot;
 		struct ast_vm_msg_snapshot *msg;
-		int i = 0;
 		mailbox = a->argv[2];
 		context = a->argv[3];
 		folder = a->argv[4];
@@ -11797,6 +11835,7 @@ static char *complete_voicemail_move_message(struct ast_cli_args *a, int maxpos)
 
 		/* Take a snapshot of the mailbox and snag the individual info */
 		if ((mailbox_snapshot = ast_vm_mailbox_snapshot_create(mailbox, context, folder, 0, AST_VM_SNAPSHOT_SORT_BY_ID, 0))) {
+			int i;
 			/* we are only requesting the one folder, but we still need to know it's index */
 			for (i = 0; i < ARRAY_LEN(mailbox_folders); i++) {
 				if (!strcasecmp(mailbox_folders[i], folder)) {
@@ -12897,7 +12936,7 @@ AST_TEST_DEFINE(test_voicemail_vmuser)
 	/* language parameter seems to only be used for display in manager action */
 	static const char options_string[] = "attach=yes|attachfmt=wav49|"
 		"serveremail=someguy@digium.com|fromstring=Voicemail System|tz=central|delete=yes|saycid=yes|"
-		"sendvoicemail=yes|review=yes|tempgreetwarn=yes|messagewrap=yes|operator=yes|"
+		"sendvoicemail=yes|review=yes|tempgreetwarn=yes|messagewrap=yes|operator=yes|leaveurgent=yes|"
 		"envelope=yes|moveheard=yes|sayduration=yes|saydurationm=5|forcename=yes|"
 		"forcegreetings=yes|callback=somecontext|dialout=somecontext2|"
 		"exitcontext=somecontext3|minsecs=10|maxsecs=100|nextaftercmd=yes|"
@@ -12973,6 +13012,10 @@ AST_TEST_DEFINE(test_voicemail_vmuser)
 		ast_test_status_update(test, "Parse failure for review option\n");
 		res = 1;
 	}
+	if (!ast_test_flag(vmu, VM_MARK_URGENT)) {
+		ast_test_status_update(test, "Parse failure for leaveurgent option\n");
+		res = 1;
+	}
 	if (!ast_test_flag(vmu, VM_TEMPGREETWARN)) {
 		ast_test_status_update(test, "Parse failure for tempgreetwarm option\n");
 		res = 1;
@@ -13079,6 +13122,7 @@ AST_TEST_DEFINE(test_voicemail_vmuser)
 #endif
 
 	free_user(vmu);
+	force_reload_config(); /* Restore original config */
 	return res ? AST_TEST_FAIL : AST_TEST_PASS;
 }
 #endif
@@ -13703,6 +13747,7 @@ static int append_vmu_info_astman(
 		"DeleteMessage: %s\r\n"
 		"VolumeGain: %.2f\r\n"
 		"CanReview: %s\r\n"
+		"CanMarkUrgent: %s\r\n"
 		"CallOperator: %s\r\n"
 		"MaxMessageCount: %d\r\n"
 		"MaxMessageLength: %d\r\n"
@@ -13740,6 +13785,7 @@ static int append_vmu_info_astman(
 		ast_test_flag(vmu, VM_DELETE) ? "Yes" : "No",
 		vmu->volgain,
 		ast_test_flag(vmu, VM_REVIEW) ? "Yes" : "No",
+		ast_test_flag(vmu, VM_MARK_URGENT) ? "Yes" : "No",
 		ast_test_flag(vmu, VM_OPERATOR) ? "Yes" : "No",
 		vmu->maxmsg,
 		vmu->maxsecs,
@@ -13773,6 +13819,7 @@ static int append_vmbox_info_astman(
 	struct ast_vm_mailbox_snapshot *mailbox_snapshot;
 	struct ast_vm_msg_snapshot *msg;
 	int nummessages = 0;
+	int i;
 
 	/* Take a snapshot of the mailbox */
 	mailbox_snapshot = ast_vm_mailbox_snapshot_create(vmu->mailbox, vmu->context, NULL, 0, AST_VM_SNAPSHOT_SORT_BY_ID, 0);
@@ -13784,7 +13831,7 @@ static int append_vmbox_info_astman(
 
 	astman_send_listack(s, m, "Voicemail box detail will follow", "start");
 	/* walk through each folder's contents and append info for each message */
-	for (int i = 0; i < mailbox_snapshot->folders; i++) {
+	for (i = 0; i < mailbox_snapshot->folders; i++) {
 		AST_LIST_TRAVERSE(&((mailbox_snapshot)->snapshots[i]), msg, msg) {
 			astman_append(s,
 				"Event: %s\r\n"
@@ -14168,10 +14215,10 @@ static const char *substitute_escapes(const char *value)
 	return ast_str_buffer(str);
 }
 
-static int load_config(int reload)
+static int load_config_force(int reload, int force)
 {
 	struct ast_config *cfg, *ucfg;
-	struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 };
+	struct ast_flags config_flags = { reload && !force ? CONFIG_FLAG_FILEUNCHANGED : 0 };
 	int res;
 
 	ast_unload_realtime("voicemail");
@@ -14209,6 +14256,11 @@ static int load_config(int reload)
 	return res;
 }
 
+static int load_config(int reload)
+{
+	return load_config_force(reload, 0);
+}
+
 #ifdef TEST_FRAMEWORK
 static int load_config_from_memory(int reload, struct ast_config *cfg, struct ast_config *ucfg)
 {
@@ -14703,6 +14755,14 @@ static int actual_load_config(int reload, struct ast_config *cfg, struct ast_con
 		}
 		ast_set2_flag((&globalflags), ast_true(val), VM_REVIEW);
 
+		if (!(val = ast_variable_retrieve(cfg, "general", "leaveurgent"))){
+			val = "yes";
+		} else if (ast_false(val)) {
+			ast_debug(1, "VM leave urgent messages disabled globally\n");
+			val = "no";
+		}
+		ast_set2_flag((&globalflags), ast_true(val), VM_MARK_URGENT);
+
 		/* Temporary greeting reminder */
 		if (!(val = ast_variable_retrieve(cfg, "general", "tempgreetwarn"))) {
 			ast_debug(1, "VM Temporary Greeting Reminder Option disabled globally\n");
@@ -15375,6 +15435,7 @@ AST_TEST_DEFINE(test_voicemail_msgcount)
 	}
 
 	free_user(vmu);
+	force_reload_config(); /* Restore original config */
 	return res;
 }
 
@@ -15485,6 +15546,7 @@ AST_TEST_DEFINE(test_voicemail_notify_endl)
 	}
 	fclose(file);
 	free_user(vmu);
+	force_reload_config(); /* Restore original config */
 	return res;
 }
 
@@ -15557,8 +15619,8 @@ AST_TEST_DEFINE(test_voicemail_load_config)
 
 #undef CHECK
 
-	/* restore config */
-	load_config(1); /* this might say "Failed to load configuration file." */
+	/* Forcibly restore the original config, to reinitialize after test */
+	force_reload_config(); /* this might say "Failed to load configuration file." */
 
 cleanup:
 	unlink(config_filename);
@@ -15624,6 +15686,11 @@ AST_TEST_DEFINE(test_voicemail_vm_info)
 	populate_defaults(vmu);
 
 	vmu->email = ast_strdup("vm-info-test@example.net");
+	if (!vmu->email) {
+		ast_test_status_update(test, "Cannot create vmu email\n");
+		chan = ast_channel_unref(chan);
+		return AST_TEST_FAIL;
+	}
 	ast_copy_string(vmu->fullname, "Test Framework Mailbox", sizeof(vmu->fullname));
 	ast_copy_string(vmu->pager, "vm-info-pager-test@example.net", sizeof(vmu->pager));
 	ast_copy_string(vmu->language, "en", sizeof(vmu->language));
@@ -16270,7 +16337,7 @@ static int play_record_review(struct ast_channel *chan, char *playfile, char *re
 			}
 			break;
 		case '4':
-			if (outsidecaller) {  /* only mark vm messages */
+			if (outsidecaller && ast_test_flag(vmu, VM_MARK_URGENT)) {  /* only mark vm messages */
 				/* Mark Urgent */
 				if ((flag && ast_strlen_zero(flag)) || (!ast_strlen_zero(flag) && strcmp(flag, "Urgent"))) {
 					ast_verb(3, "marking message as Urgent\n");
@@ -16347,7 +16414,7 @@ static int play_record_review(struct ast_channel *chan, char *playfile, char *re
 				return cmd;
 			if (msg_exists) {
 				cmd = ast_play_and_wait(chan, "vm-review");
-				if (!cmd && outsidecaller) {
+				if (!cmd && outsidecaller && ast_test_flag(vmu, VM_MARK_URGENT)) {
 					if ((flag && ast_strlen_zero(flag)) || (!ast_strlen_zero(flag) && strcmp(flag, "Urgent"))) {
 						cmd = ast_play_and_wait(chan, "vm-review-urgent");
 					} else if (flag) {

apps/confbridge/conf_config_parser.c

@@ -2220,6 +2220,30 @@ static int user_template_handler(const struct aco_option *opt, struct ast_variab
 	return conf_find_user_profile(NULL, var->value, u_profile) ? 0 : -1;
 }
 
+static int sample_rate_handler(const struct aco_option *opt, struct ast_variable *var, void *obj)
+{
+	struct bridge_profile *b_profile = obj;
+	unsigned int *slot;
+
+	if (!strcasecmp(var->name, "internal_sample_rate")) {
+		slot = &b_profile->internal_sample_rate;
+		if (!strcasecmp(var->value, "auto")) {
+			*slot = 0;
+			return 0;
+		}
+	} else if (!strcasecmp(var->name, "maximum_sample_rate")) {
+		slot = &b_profile->maximum_sample_rate;
+		if (!strcasecmp(var->value, "none")) {
+			*slot = 0;
+			return 0;
+		}
+	} else {
+		return -1;
+	}
+
+	return ast_parse_arg(var->value, PARSE_UINT32 | PARSE_IN_RANGE, slot, 8000, 192000);
+}
+
 static int bridge_template_handler(const struct aco_option *opt, struct ast_variable *var, void *obj)
 {
 	struct bridge_profile *b_profile = obj;
@@ -2437,10 +2461,9 @@ int conf_load_config(void)
 	/* Bridge options */
 	aco_option_register(&cfg_info, "type", ACO_EXACT, bridge_types, NULL, OPT_NOOP_T, 0, 0);
 	aco_option_register(&cfg_info, "jitterbuffer", ACO_EXACT, bridge_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct bridge_profile, flags), USER_OPT_JITTERBUFFER);
-	/* "auto" will fail to parse as a uint, but we use PARSE_DEFAULT to set the value to 0 in that case, which is the value that auto resolves to */
-	aco_option_register(&cfg_info, "internal_sample_rate", ACO_EXACT, bridge_types, "0", OPT_UINT_T, PARSE_DEFAULT, FLDSET(struct bridge_profile, internal_sample_rate), 0);
+	aco_option_register_custom(&cfg_info, "internal_sample_rate", ACO_EXACT, bridge_types, "auto", sample_rate_handler, 0);
 	aco_option_register(&cfg_info, "binaural_active", ACO_EXACT, bridge_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct bridge_profile, flags), BRIDGE_OPT_BINAURAL_ACTIVE);
-	aco_option_register(&cfg_info, "maximum_sample_rate", ACO_EXACT, bridge_types, "0", OPT_UINT_T, PARSE_DEFAULT, FLDSET(struct bridge_profile, maximum_sample_rate), 0);
+	aco_option_register_custom(&cfg_info, "maximum_sample_rate", ACO_EXACT, bridge_types, "none", sample_rate_handler, 0);
 	aco_option_register_custom(&cfg_info, "mixing_interval", ACO_EXACT, bridge_types, "20", mix_interval_handler, 0);
 	aco_option_register(&cfg_info, "record_conference", ACO_EXACT, bridge_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct bridge_profile, flags), BRIDGE_OPT_RECORD_CONFERENCE);
 	aco_option_register_custom(&cfg_info, "video_mode", ACO_EXACT, bridge_types, NULL, video_mode_handler, 0);

apps/confbridge/include/conf_state.h

@@ -22,7 +22,7 @@
  *
  * \author\verbatim Terry Wilson <twilson@digium.com> \endverbatim
  *
- * See https://wiki.asterisk.org/wiki/display/AST/Confbridge+state+changes for
+ * See https://docs.asterisk.org/Development/Reference-Information/Other-Reference-Information/Confbridge-state-changes/ for
  * a more complete description of how conference states work.
  */
 

bootstrap.sh

@@ -9,9 +9,9 @@ check_for_app() {
 	fi
 }
 
-# OpenBSD: pkg_add autoconf%2.63 automake%1.9 metaauto
-test -n "$AUTOCONF_VERSION" || export AUTOCONF_VERSION=2.63
-test -n "$AUTOMAKE_VERSION" || export AUTOMAKE_VERSION=1.9
+# OpenBSD: pkg_add autoconf%2.69 automake%1.16 metaauto
+test -n "$AUTOCONF_VERSION" || export AUTOCONF_VERSION=2.69
+test -n "$AUTOMAKE_VERSION" || export AUTOMAKE_VERSION=1.16
 
 check_for_app autoconf
 check_for_app autoheader

bridges/bridge_simple.c

@@ -181,7 +181,14 @@ static int simple_bridge_join(struct ast_bridge *bridge, struct ast_bridge_chann
 		return 0;
 	}
 
-	ast_channel_request_stream_topology_change(c1, new_top, &simple_bridge);
+	if (!ast_stream_topology_equal(new_top, existing_top)) {
+		ast_channel_request_stream_topology_change(c1, new_top, &simple_bridge);
+	} else {
+		ast_debug(3, "%s: Topologies already match. Current: %s  Requested: %s\n",
+				ast_channel_name(c1),
+				ast_str_tmp(256, ast_stream_topology_to_str(existing_top, &STR_TMP)),
+				ast_str_tmp(256, ast_stream_topology_to_str(new_top, &STR_TMP)));
+	}
 	ast_stream_topology_free(new_top);
 
 	return 0;

build_tools/cflags-devmode.xml

@@ -9,7 +9,6 @@
 			<support_level>extended</support_level>
 		</member>
 		<member name="DETECT_DEADLOCKS" displayname="Detect Deadlocks">
-			<depend>DEBUG_THREADS</depend>
 			<support_level>extended</support_level>
 		</member>
 		<member name="DUMP_SCHEDULER" displayname="Dump Scheduler Contents for Debugging">

build_tools/cflags.xml

@@ -128,4 +128,8 @@
 			<defaultenabled>yes</defaultenabled>
 			<depend>native_arch</depend>
 		</member>
+		<member name="ADD_CFLAGS_TO_BUILDOPTS_H" displayname="Add ALL of the flags on this page to buildopts.h. Useful for IDEs but may cause slightly longer compile times after flags are changed.">
+			<support_level>core</support_level>
+			<defaultenabled>no</defaultenabled>
+		</member>
 	</category>

build_tools/download_externals

@@ -58,7 +58,7 @@ if [[ -z ${cache_dir} ]] ; then
 fi
 
 version=$(${ASTTOPDIR}/build_tools/make_version ${ASTTOPDIR})
-if [[ ! ${version} =~ ^(GIT-)?(certified/)?([^.-]+)[.-].* ]] ; then
+if [[ ! ${version} =~ ^(GIT-)?(certified[/-])?([^.-]+)[.-].* ]] ; then
 	echo "${module_name}: Couldn't parse version ${version}"
 	exit 1
 fi
@@ -172,7 +172,7 @@ if [[ -f ${DESTDIR}${ASTMODDIR}/${module_name}.manifest.xml ]] ; then
 
 				cs=$(${MD5} ${f} | cut -b1-32)
 				if [[ "${cs}" !=  "${sum}" ]] ; then
-					echo Checksum mismatch: ${f}
+					echo "Checksum mismatch: ${f}"
 					need_install=1
 					break
 				fi
@@ -194,8 +194,8 @@ else
 fi
 
 need_download=1
-if [[ -f ${cache_dir}/${full_name}.manifest.xml ]] ; then
-	cpv=$(${XMLSTARLET} sel -t -v "/package/@version" ${cache_dir}/${full_name}.manifest.xml)
+if [[ -f ${cache_dir}/${full_name}-${major_version}.manifest.xml ]] ; then
+	cpv=$(${XMLSTARLET} sel -t -v "/package/@version" ${cache_dir}/${full_name}-${major_version}.manifest.xml)
 	cpvi=$(version_convert ${cpv})
 	echo "${full_name}: Cached package version ${cpv} (${cpvi})"
 	if [[ ${cpvi} == ${rpvi} && ( -f ${cache_dir}/${tarball} ) ]] ; then
@@ -210,7 +210,7 @@ if [[ ${need_download} = 1 ]] ; then
 		echo "${full_name}: Unable to fetch ${remote_url}/${tarball}"
 		exit 1
 	}
-	cp ${tmpdir}/${variant_manifest}  ${cache_dir}/${full_name}.manifest.xml
+	cp ${tmpdir}/${variant_manifest}  ${cache_dir}/${full_name}-${major_version}.manifest.xml
 fi
 
 tar -xzf ${cache_dir}/${tarball} -C ${cache_dir}

build_tools/make_buildopts_h

@@ -18,42 +18,79 @@ then
 	# gets added to BUILDOPTS.
 fi
 
-TMP=`${GREP} -e "^MENUSELECT_CFLAGS" menuselect.makeopts | sed 's/MENUSELECT_CFLAGS\=//g' | sed 's/-D//g'`
-for x in ${TMP}; do
-	if test "${x}" = "AO2_DEBUG" \
-			-o "${x}" = "BETTER_BACKTRACES" \
-			-o "${x}" = "BUILD_NATIVE" \
-			-o "${x}" = "COMPILE_DOUBLE" \
-			-o "${x}" = "DEBUG_CHAOS" \
-			-o "${x}" = "DEBUG_SCHEDULER" \
-			-o "${x}" = "DETECT_DEADLOCKS" \
-			-o "${x}" = "DONT_OPTIMIZE" \
-			-o "${x}" = "DUMP_SCHEDULER" \
-			-o "${x}" = "LOTS_OF_SPANS" \
-			-o "${x}" = "MALLOC_DEBUG" \
-			-o "${x}" = "RADIO_RELAX" \
-			-o "${x}" = "REBUILD_PARSERS" \
-			-o "${x}" = "REF_DEBUG" \
-			-o "${x}" = "USE_HOARD_ALLOCATOR" ; then
-		# These options are only for specific sources and have no effect on public ABI.
-		# Keep them out of buildopts.h so ccache does not invalidate all sources.
-		continue
-	fi
+ADD_CFLAGS_TO_BUILDOPTS=false
+MENUSELECT_CFLAGS=$(${GREP} -e "^MENUSELECT_CFLAGS" menuselect.makeopts)
+echo "$MENUSELECT_CFLAGS" | grep -q -e "ADD_CFLAGS_TO_BUILDOPTS_H" && ADD_CFLAGS_TO_BUILDOPTS=true
 
+# Clean up MENUSELECT_CFLAGS by removing the "MENUSELECT_CFLAGS="
+# at the front, the "ADD_CFLAGS_TO_BUILDOPTS_H" flag, and any "-D"
+# entries.
+MENUSELECT_CFLAGS=$( echo "$MENUSELECT_CFLAGS" | \
+	sed -r -e "s/(MENUSELECT_CFLAGS=|ADD_CFLAGS_TO_BUILDOPTS_H|-D)//g")
+
+# This is a list of flags that don't affect the ABI.
+# "ADD_CFLAGS_TO_BUILDOPTS_H" is NOT set, we'll filter these
+# out of the buildopts.h file.
+#
+# These used to always be filtered out but if they're not in
+# buildopts.h, many IDEs will show them as undefined and mark
+# any code blocks enabled by them as disabled.
+#
+# The original reasoning for removing them was that trivial
+# changes to the buildopts.h file will cause ccache to
+# invalidate any source files that use it and increase the
+# compile time.  It's not such a huge deal these days but
+# to preserve backwards behavior the default is still to
+# remove them.
+# 
+# The ABI-breaking flags are always included in buildopts.h.
+
+# This variable is used by sed so it needs to be a valid
+# regex which will be surrounded by parens.]
+FILTER_OUT="\
+AO2_DEBUG|BETTER_BACKTRACES|BUILD_NATIVE|\
+COMPILE_DOUBLE|DEBUG_CHAOS|DEBUG_SCHEDULER|\
+DETECT_DEADLOCKS|DONT_OPTIMIZE|DUMP_SCHEDULER|\
+LOTS_OF_SPANS|MALLOC_DEBUG|RADIO_RELAX|\
+REBUILD_PARSERS|REF_DEBUG|USE_HOARD_ALLOCATOR"
+
+# Create buildopts.h
+
+INCLUDE_CFLAGS="$MENUSELECT_CFLAGS"
+# Do the filter-out if needed.
+if ! $ADD_CFLAGS_TO_BUILDOPTS ; then
+	INCLUDE_CFLAGS=$( echo "$MENUSELECT_CFLAGS" | \
+		sed -r -e "s/(${FILTER_OUT})//g")
+fi
+
+# Output the defines.
+for x in ${INCLUDE_CFLAGS}; do
 	echo "#define ${x} 1"
-	if test "${x}" = "LOW_MEMORY" ; then
-		# LOW_MEMORY isn't an ABI affecting option but it is used in many sources
-		# so it gets defined globally but is not included in AST_BUILTOPTS.
-		continue
-	fi
-	if test "x${BUILDOPTS}" != "x" ; then
-		BUILDOPTS="${BUILDOPTS}, ${x}"
-	else
-		BUILDOPTS="${x}"
-	fi
 done
 
-BUILDSUM=`echo ${BUILDOPTS} | ${MD5} | cut -c1-32`
+# We NEVER include the non-ABI-breaking flags in the
+# BUILDOPTS or use them to calculate the checksum so
+# we always filter out any that may exist.
+# After the filter-out, we also need to convert the
+# possibly-multi-spaced MENUSELECT_CFLAGS to a nice
+# comma-separated list.
+# I.E.
+#  Remove leading spaces.
+#  Convert consecutive interior spaces to a single space.
+#  Remove trailing spaces.
+#  Convert the now-single-spaces in the interior to ", ".
+BUILDOPTS=$( echo "$MENUSELECT_CFLAGS" | \
+	sed -r -e "s/(${FILTER_OUT}|LOW_MEMORY)//g" -e "s/^\s+//g;s/\s+/ /g;s/\s+$//g;s/\s/, /g" )
+
+# Calculate the checksum on only the ABI-breaking flags.
+BUILDSUM=$(echo "${BUILDOPTS}" | ${MD5} | cut -c1-32)
 
 echo "#define AST_BUILDOPT_SUM \"${BUILDSUM}\""
 echo "#define AST_BUILDOPTS \"${BUILDOPTS}\""
+
+# However, it'd be nice to see the non-ABI-breaking flags
+# when you do a "core show settings" so we create a separate
+# define for them.
+BUILDOPTS_ALL=$( echo "$MENUSELECT_CFLAGS" | \
+	sed -r -e "s/^\s+//g;s/\s+/ /g;s/\s+$//g;s/\s/, /g" )
+echo "#define AST_BUILDOPTS_ALL \"${BUILDOPTS_ALL}\""

build_tools/make_version_c

@@ -2,6 +2,11 @@
 
 GREP=${GREP:-grep}
 
+if test ! -f include/asterisk/buildopts.h ; then
+	echo "include/asterisk/buildopts.h is missing"
+	exit 1
+fi
+
 if test ! -f .flavor ; then
     EXTRA=""
 elif test ! -f .version ; then
@@ -18,14 +23,11 @@ then
 	BUILDOPTS="AST_DEVMODE"
 fi
 
-TMP=`${GREP} -e "^MENUSELECT_CFLAGS" menuselect.makeopts | sed 's/MENUSELECT_CFLAGS\=//g' | sed 's/-D//g'`
-for x in ${TMP}; do
-	if test "x${BUILDOPTS}" != "x" ; then
-		BUILDOPTS="${BUILDOPTS}, ${x}"
-	else
-		BUILDOPTS="${x}"
-	fi
-done
+BUILDOPTS=$(sed -n -r -e 's/#define\s+AST_BUILDOPTS\s+"([^"]+)"/\1/gp' \
+	include/asterisk/buildopts.h )
+
+BUILDOPTS_ALL=$(sed -n -r -e 's/#define\s+AST_BUILDOPTS_ALL\s+"([^"]+)"/\1/gp' \
+	include/asterisk/buildopts.h )
 
 cat << END
 /*
@@ -43,6 +45,8 @@ static const char asterisk_version_num[] = "${ASTERISKVERSIONNUM}";
 
 static const char asterisk_build_opts[] = "${BUILDOPTS}";
 
+static const char asterisk_build_opts_all[] = "${BUILDOPTS_ALL}";
+
 const char *ast_get_version(void)
 {
 	return asterisk_version;
@@ -58,4 +62,9 @@ const char *ast_get_build_opts(void)
 	return asterisk_build_opts;
 }
 
+const char *ast_get_build_opts_all(void)
+{
+	return asterisk_build_opts_all;
+}
+
 END

build_tools/make_xml_documentation

@@ -135,12 +135,18 @@ if [ "${for_wiki}" -eq "1" ] || [ "${validate}" -eq "1" ]; then
 	fi
 fi
 
+make_absolute() {
+	case "$1" in
+		/*) echo "$1" ;;
+		*)  echo "$source_tree/$1" ;;
+	esac
+}
+
 if [ "${command}" = "print_dependencies" ] ; then
 	for subdir in ${mod_subdirs} ; do
-		subpath="${source_tree}/${subdir}"
-		# We WANT word splitting in the following line.
-		# shellcheck disable=SC2046
-		${GREP} -l -E '(language="en_US"|appdocsxml.dtd)' $(${FIND} "${subpath}" -name '*.c' -or -name '*.cc' -or -name '*.xml') || :
+		subpath=$(make_absolute "$subdir")
+		${FIND} "${subpath}" \( -name '*.c' -o -name '*.cc' -o -name '*.xml' \) \
+			-exec ${GREP} -l -E '(language="en_US"|appdocsxml.dtd)' '{}' \;
 	done
 	exit
 fi
@@ -186,7 +192,7 @@ printf "Building Documentation For: "
 
 for subdir in ${mod_subdirs} ; do
 	printf "%s " "${subdir}"
-	subdir_path="${source_tree}/${subdir}"
+	subdir_path=$(make_absolute "$subdir")
 	for i in $(${FIND} "${subdir_path}" -name '*.c' -or -name '*.cc'); do
 		if [ "${with_moduleinfo}" -eq "1" ] ; then
 			MODULEINFO=$(${AWK} -f "${source_tree}/build_tools/get_moduleinfo" "${i}")

build_tools/menuselect-deps.in

@@ -28,6 +28,7 @@ URIPARSER=@PBX_URIPARSER@
 KQUEUE=@PBX_KQUEUE@
 LDAP=@PBX_LDAP@
 LIBEDIT=@PBX_LIBEDIT@
+LIBJWT=@PBX_LIBJWT@
 LIBXML2=@PBX_LIBXML2@
 LIBXSLT=@PBX_LIBXSLT@
 XMLSTARLET=@PBX_XMLSTARLET@

channels/chan_console.c

@@ -152,6 +152,8 @@ static struct console_pvt {
 	struct ast_frame fr;
 	/*! Running = 1, Not running = 0 */
 	unsigned int streamstate:1;
+	/*! Abort stream processing? */
+	unsigned int abort:1;
 	/*! On-hook = 0, Off-hook = 1 */
 	unsigned int hookstate:1;
 	/*! Unmuted = 0, Muted = 1 */
@@ -275,18 +277,19 @@ static void *stream_monitor(void *data)
 	};
 
 	for (;;) {
-		pthread_testcancel();
 		console_pvt_lock(pvt);
 		res = Pa_ReadStream(pvt->stream, buf, sizeof(buf) / sizeof(int16_t));
 		console_pvt_unlock(pvt);
-		pthread_testcancel();
 
-		if (!pvt->owner) {
+		if (!pvt->owner || pvt->abort) {
 			return NULL;
 		}
 
-		if (res == paNoError)
+		if (res == paNoError) {
 			ast_queue_frame(pvt->owner, &f);
+		} else {
+			ast_log(LOG_WARNING, "Console ReadStream failed: %s\n", Pa_GetErrorText(res));
+		}
 	}
 
 	return NULL;
@@ -401,8 +404,9 @@ static int stop_stream(struct console_pvt *pvt)
 	if (!pvt->streamstate || pvt->thread == AST_PTHREADT_NULL)
 		return 0;
 
-	pthread_cancel(pvt->thread);
-	pthread_kill(pvt->thread, SIGURG);
+	pvt->abort = 1;
+	/* Wait for pvt->thread to exit cleanly, to avoid killing it while it's holding a lock. */
+	pthread_kill(pvt->thread, SIGURG); /* Wake it up if needed, but don't cancel it */
 	pthread_join(pvt->thread, NULL);
 
 	console_pvt_lock(pvt);

channels/chan_dahdi.c

@@ -263,6 +263,10 @@
 			</enum>
 			<enum name="dialmode">
 				<para>R/W Pulse and tone dialing mode of the channel.</para>
+				<para>Disabling tone dialing using this option will not disable the DSP used for DTMF detection.
+				To do that, also set the <literal>digitdetect</literal> option. If digit detection is disabled,
+				DTMF will not be detected, regardless of the <literal>dialmode</literal> setting.
+				The <literal>digitdetect</literal> setting has no impact on pulse dialing detection.</para>
 				<para>If set, overrides the setting in <literal>chan_dahdi.conf</literal> for that channel.</para>
 				<enumlist>
 					<enum name="both" />
@@ -2024,6 +2028,9 @@ static void my_set_cadence(void *pvt, int *cid_rings, struct ast_channel *ast)
 			ast_log(LOG_WARNING, "Unable to set distinctive ring cadence %d on '%s': %s\n", p->distinctivering, ast_channel_name(ast), strerror(errno));
 		*cid_rings = cidrings[p->distinctivering - 1];
 	} else {
+		if (p->distinctivering > 0) {
+			ast_log(LOG_WARNING, "Cadence %d is not defined, falling back to default ring cadence\n", p->distinctivering);
+		}
 		if (ioctl(p->subs[SUB_REAL].dfd, DAHDI_SETCADENCE, NULL))
 			ast_log(LOG_WARNING, "Unable to reset default ring on '%s': %s\n", ast_channel_name(ast), strerror(errno));
 		*cid_rings = p->sendcalleridafter;
@@ -5223,6 +5230,18 @@ static int has_voicemail(struct dahdi_pvt *p)
 	int new_msgs;
 	RAII_VAR(struct stasis_message *, mwi_message, NULL, ao2_cleanup);
 
+	/* A manual MWI disposition has been requested, use that instead
+	 * if this is for sending the new MWI indication. */
+	if (p->mwioverride_active) {
+		/* We don't clear p->mwioverride_active automatically,
+		 * because otherwise do_monitor would just change it back to the way it was.
+		 * We need to keep the override active until explicitly disabled by the user,
+		 * so that we can keep returning the correct answer in subsequent calls to do_monitor. */
+		ast_debug(6, "MWI manual override active on channel %d: pretending that it should be %s\n",
+			p->channel, p->mwioverride_disposition ? "active" : "inactive");
+		return p->mwioverride_disposition;
+	}
+
 	mwi_message = stasis_cache_get(ast_mwi_state_cache(), ast_mwi_state_type(), p->mailbox);
 	if (mwi_message) {
 		struct ast_mwi_state *mwi_state = stasis_message_data(mwi_message);
@@ -6719,6 +6738,14 @@ static int dahdi_queryoption(struct ast_channel *chan, int option, void *data, i
 	}
 
 	switch (option) {
+	case AST_OPTION_TDD:
+		cp = (char *) data;
+		if (p->mate) {
+			*cp = 2;
+		} else {
+			*cp = p->tdd ? 1 : 0;
+		}
+		break;
 	case AST_OPTION_DIGIT_DETECT:
 		cp = (char *) data;
 		*cp = p->ignoredtmf ? 0 : 1;
@@ -11874,7 +11901,7 @@ static void *do_monitor(void *data)
 							&& (last->sig & __DAHDI_SIG_FXO)
 							&& !analog_p->fxsoffhookstate
 							&& !last->owner
-							&& !ast_strlen_zero(last->mailbox)
+							&& (!ast_strlen_zero(last->mailbox) || last->mwioverride_active)
 							&& !analog_p->subs[SUB_REAL].owner /* could be a recall ring from a flash hook hold */
 							&& (thispass - analog_p->onhooktime > 3)) {
 							res = has_voicemail(last);
@@ -12949,6 +12976,7 @@ static struct dahdi_pvt *mkintf(int channel, const struct dahdi_chan_conf *conf,
 		tmp->callwaitingcallerid = conf->chan.callwaitingcallerid;
 		tmp->threewaycalling = conf->chan.threewaycalling;
 		tmp->threewaysilenthold = conf->chan.threewaysilenthold;
+		tmp->calledsubscriberheld = conf->chan.calledsubscriberheld; /* Not used in chan_dahdi.c, just analog pvt, but must exist on the DAHDI pvt anyways */
 		tmp->adsi = conf->chan.adsi;
 		tmp->use_smdi = conf->chan.use_smdi;
 		tmp->permhidecallerid = conf->chan.hidecallerid;
@@ -13247,6 +13275,7 @@ static struct dahdi_pvt *mkintf(int channel, const struct dahdi_chan_conf *conf,
 				analog_p->ani_wink_time = conf->chan.ani_wink_time;
 				analog_p->hanguponpolarityswitch = conf->chan.hanguponpolarityswitch;
 				analog_p->permcallwaiting = conf->chan.callwaiting; /* permcallwaiting possibly modified in analog_config_complete */
+				analog_p->calledsubscriberheld = conf->chan.calledsubscriberheld; /* Only actually used in analog pvt, not DAHDI pvt */
 				analog_p->callreturn = conf->chan.callreturn;
 				analog_p->cancallforward = conf->chan.cancallforward;
 				analog_p->canpark = conf->chan.canpark;
@@ -16524,6 +16553,75 @@ static char *dahdi_set_dnd(struct ast_cli_entry *e, int cmd, struct ast_cli_args
 	return CLI_SUCCESS;
 }
 
+static char *dahdi_set_mwi(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+{
+	int channel;
+	int on;
+	int override = 1;
+	struct dahdi_pvt *dahdi_chan = NULL;
+
+	switch (cmd) {
+	case CLI_INIT:
+		e->command = "dahdi set mwi";
+		e->usage =
+			"Usage: dahdi set mwi <chan#> <on|off|reset>\n"
+			"	Sets/unsets MWI (Message Waiting Indicator) manually on a channel.\n"
+			"   This may be used regardless of whether the channel is assigned any mailboxes.\n"
+			"   When active, this setting will override the voicemail status to set MWI.\n"
+			"   Once cleared, the voicemail status will resume control of MWI.\n"
+			"	Changes are queued for when the channel is idle and persist until cleared.\n"
+			"	<chan num> is the channel number\n"
+			" 	<on|off|reset> Enable, disable, or reset Message Waiting Indicator override?\n"
+			;
+		return NULL;
+	case CLI_GENERATE:
+		return NULL;
+	}
+
+	if (a->argc != 5)
+		return CLI_SHOWUSAGE;
+
+	if ((channel = atoi(a->argv[3])) <= 0) {
+		ast_cli(a->fd, "Expected channel number, got '%s'\n", a->argv[3]);
+		return CLI_SHOWUSAGE;
+	}
+
+	if (ast_true(a->argv[4])) {
+		on = 1;
+	} else if (ast_false(a->argv[4])) {
+		on = 0;
+	} else if (!strcmp(a->argv[4], "reset")) {
+		override = 0;
+	} else {
+		ast_cli(a->fd, "Expected 'on' or 'off' or 'reset', got '%s'\n", a->argv[4]);
+		return CLI_SHOWUSAGE;
+	}
+
+	ast_mutex_lock(&iflock);
+	for (dahdi_chan = iflist; dahdi_chan; dahdi_chan = dahdi_chan->next) {
+		if (dahdi_chan->channel != channel)
+			continue;
+
+		/* Found the channel. Actually set it */
+		if (override) {
+			dahdi_chan->mwioverride_disposition = on;
+			ast_cli(a->fd, "MWI '%s' queued for channel %d\n", on ? "enable" : "disable", channel);
+		}
+		dahdi_chan->mwioverride_active = override;
+		/* The do_monitor thread will take care of actually sending the MWI
+		 * at an appropriate time for the channel. */
+		break;
+	}
+	ast_mutex_unlock(&iflock);
+
+	if (!dahdi_chan) {
+		ast_cli(a->fd, "Unable to find given channel %d\n", channel);
+		return CLI_FAILURE;
+	}
+
+	return CLI_SUCCESS;
+}
+
 static struct ast_cli_entry dahdi_cli[] = {
 	AST_CLI_DEFINE(handle_dahdi_show_cadences, "List cadences"),
 	AST_CLI_DEFINE(dahdi_show_channels, "Show active DAHDI channels"),
@@ -16536,6 +16634,7 @@ static struct ast_cli_entry dahdi_cli[] = {
 	AST_CLI_DEFINE(dahdi_set_hwgain, "Set hardware gain on a channel"),
 	AST_CLI_DEFINE(dahdi_set_swgain, "Set software gain on a channel"),
 	AST_CLI_DEFINE(dahdi_set_dnd, "Sets/resets DND (Do Not Disturb) mode on a channel"),
+	AST_CLI_DEFINE(dahdi_set_mwi, "Sets/unsets MWI (Message Waiting Indicator) manually on a channel"),
 };
 
 #define TRANSFER	0
@@ -18341,6 +18440,8 @@ static int process_dahdi(struct dahdi_chan_conf *confp, const char *cat, struct
 			confp->chan.busycount = atoi(v->value);
 		} else if (!strcasecmp(v->name, "busypattern")) {
 			parse_busy_pattern(v, &confp->chan.busy_cadence);
+		} else if (!strcasecmp(v->name, "calledsubscriberheld")) {
+			confp->chan.calledsubscriberheld = ast_true(v->value);
 		} else if (!strcasecmp(v->name, "callprogress")) {
 			confp->chan.callprogress &= ~CALLPROGRESS_PROGRESS;
 			if (ast_true(v->value))
@@ -18430,18 +18531,30 @@ static int process_dahdi(struct dahdi_chan_conf *confp, const char *cat, struct
 		} else if (!strcasecmp(v->name, "group")) {
 			confp->chan.group = ast_get_group(v->value);
 		} else if (!strcasecmp(v->name, "callgroup")) {
+			if (!((confp->chan.sig == SIG_FXOKS) || (confp->chan.sig == SIG_FXOGS) || (confp->chan.sig == SIG_FXOLS))) {
+				ast_log(LOG_WARNING, "Only FXO signalled channels may belong to a call group\n");
+			}
 			if (!strcasecmp(v->value, "none"))
 				confp->chan.callgroup = 0;
 			else
 				confp->chan.callgroup = ast_get_group(v->value);
 		} else if (!strcasecmp(v->name, "pickupgroup")) {
+			if (!((confp->chan.sig == SIG_FXOKS) || (confp->chan.sig == SIG_FXOGS) || (confp->chan.sig == SIG_FXOLS))) {
+				ast_log(LOG_WARNING, "Only FXO signalled channels may belong to a pickup group\n");
+			}
 			if (!strcasecmp(v->value, "none"))
 				confp->chan.pickupgroup = 0;
 			else
 				confp->chan.pickupgroup = ast_get_group(v->value);
 		} else if (!strcasecmp(v->name, "namedcallgroup")) {
+			if (!((confp->chan.sig == SIG_FXOKS) || (confp->chan.sig == SIG_FXOGS) || (confp->chan.sig == SIG_FXOLS))) {
+				ast_log(LOG_WARNING, "Only FXO signalled channels may belong to a named call group\n");
+			}
 			confp->chan.named_callgroups = ast_get_namedgroups(v->value);
 		} else if (!strcasecmp(v->name, "namedpickupgroup")) {
+			if (!((confp->chan.sig == SIG_FXOKS) || (confp->chan.sig == SIG_FXOGS) || (confp->chan.sig == SIG_FXOLS))) {
+				ast_log(LOG_WARNING, "Only FXO signalled channels may belong to a named pickup group\n");
+			}
 			confp->chan.named_pickupgroups = ast_get_namedgroups(v->value);
 		} else if (!strcasecmp(v->name, "setvar")) {
 			if (v->value) {

channels/chan_dahdi.h

@@ -204,6 +204,13 @@ struct dahdi_pvt {
 	 * \note Set from the "busydetect" value read in from chan_dahdi.conf
 	 */
 	unsigned int busydetect:1;
+	/*!
+	 * \brief TRUE if Called Subscriber held is enabled.
+	 * This allows a single incoming call to hold a DAHDI channel up,
+	 * allowing a recipient to hang up an extension and pick up another
+	 * phone on the same line without disconnecting the call.
+	 */
+	unsigned int calledsubscriberheld:1;
 	/*!
 	 * \brief TRUE if call return is enabled.
 	 * (*69, if your dialplan doesn't catch this first)
@@ -424,6 +431,10 @@ struct dahdi_pvt {
 	unsigned int mwimonitoractive:1;
 	/*! \brief TRUE if a MWI message sending thread is active */
 	unsigned int mwisendactive:1;
+	/*! \brief TRUE if a manual MWI override is active for a channel */
+	unsigned int mwioverride_active:1;
+	/*! \brief Manual MWI disposition (on/off) */
+	unsigned int mwioverride_disposition:1;
 	/*!
 	 * \brief TRUE if channel is out of reset and ready
 	 * \note Used by SS7.  Otherwise set but not used.

channels/chan_iax2.c

@@ -398,6 +398,47 @@ static int (*iax2_regfunk)(const char *username, int onoff) = NULL;
 #define DEFAULT_FREQ_OK		60 * 1000	/* How often to check for the host to be up */
 #define DEFAULT_FREQ_NOTOK	10 * 1000	/* How often to check, if the host is down... */
 
+/*! \brief Name of effective auth method */
+static const char *auth_method_labels[] = {
+    [0] = "none",
+    [IAX_AUTH_PLAINTEXT] = "plaintext",
+    [IAX_AUTH_MD5] = "MD5",
+    [IAX_AUTH_RSA] = "RSA",
+};
+
+/* Max length is length of |RSA|MD5|plaintext (18 + 1 for NUL = 19) */
+#define AUTH_METHOD_NAMES_BUFSIZE 19
+
+/*!
+ * \brief Get names of all auth methods
+ * \param Bit field of auth methods
+ * \param[out] buf Buffer into which to write the names. Must be of size AUTH_METHOD_NAMES_BUFSIZE.
+ * \return Auth methods name
+ */
+static char *auth_method_names(int authmethods, char *restrict buf)
+{
+	char *pos = buf;
+
+	*pos = '\0';
+
+	if (authmethods & IAX_AUTH_RSA) {
+		pos += sprintf(pos, "|RSA");
+	}
+	if (authmethods & IAX_AUTH_MD5) {
+		pos += sprintf(pos, "|MD5");
+	}
+	if (authmethods & IAX_AUTH_PLAINTEXT) {
+		pos += sprintf(pos, "|plaintext");
+	}
+
+	if (pos == buf) { /* No auth methods */
+		strcpy(buf, "none");
+		return buf;
+	}
+
+	return buf + 1; /* Skip leading | */
+}
+
 /* if a pvt has encryption setup done and is running on the call */
 #define IAX_CALLENCRYPTED(pvt) \
 	(ast_test_flag64(pvt, IAX_ENCRYPTED) && ast_test_flag64(pvt, IAX_KEYPOPULATED))
@@ -825,6 +866,8 @@ struct chan_iax2_pvt {
 	int authrej;
 	/*! permitted authentication methods */
 	int authmethods;
+	/*! effective authentication method */
+	int eff_auth_method;
 	/*! permitted encryption methods */
 	int encmethods;
 	/*! Encryption AES-128 Key */
@@ -8189,7 +8232,7 @@ static int authenticate_verify(struct chan_iax2_pvt *p, struct iax_ies *ies)
 		user = user_unref(user);
 	}
 	if (ast_test_flag64(p, IAX_FORCE_ENCRYPT) && !p->encmethods) {
-		ast_log(LOG_NOTICE, "Call Terminated, Incoming call is unencrypted while force encrypt is enabled.\n");
+		ast_log(LOG_WARNING, "Call Terminated, incoming call is unencrypted while force encrypt is enabled.\n");
 		return res;
 	}
 	if (!ast_test_flag(&p->state, IAX_STATE_AUTHENTICATED))
@@ -8215,12 +8258,17 @@ static int authenticate_verify(struct chan_iax2_pvt *p, struct iax_ies *ies)
 			key = ast_key_get(keyn, AST_KEY_PUBLIC);
 			if (key && !ast_check_signature(key, p->challenge, rsasecret)) {
 				res = 0;
+				p->eff_auth_method = IAX_AUTH_RSA;
 				break;
-			} else if (!key)
-				ast_log(LOG_WARNING, "requested inkey '%s' for RSA authentication does not exist\n", keyn);
+			} else if (!key) {
+				ast_log(LOG_WARNING, "Requested inkey '%s' for RSA authentication does not exist\n", keyn);
+			}
 			keyn = strsep(&stringp, ":");
 		}
 		ast_free(tmpkey);
+		if (res && authdebug) {
+			ast_log(LOG_WARNING, "No RSA public keys on file matched incoming call\n");
+		}
 	} else if (p->authmethods & IAX_AUTH_MD5) {
 		struct MD5Context md5;
 		unsigned char digest[16];
@@ -8237,12 +8285,19 @@ static int authenticate_verify(struct chan_iax2_pvt *p, struct iax_ies *ies)
 				sprintf(requeststr + (x << 1), "%02hhx", digest[x]); /* safe */
 			if (!strcasecmp(requeststr, md5secret)) {
 				res = 0;
+				p->eff_auth_method = IAX_AUTH_MD5;
 				break;
+			} else if (authdebug) {
+				ast_log(LOG_WARNING, "MD5 secret mismatch\n");
 			}
 		}
 	} else if (p->authmethods & IAX_AUTH_PLAINTEXT) {
-		if (!strcmp(secret, p->secret))
+		if (!strcmp(secret, p->secret)) {
 			res = 0;
+			p->eff_auth_method = IAX_AUTH_PLAINTEXT;
+		} else if (authdebug) {
+			ast_log(LOG_WARNING, "Plaintext secret mismatch\n");
+		}
 	}
 	return res;
 }
@@ -8417,22 +8472,25 @@ static int authenticate(const char *challenge, const char *secret, const char *k
 	if (!ast_strlen_zero(keyn)) {
 		if (!(authmethods & IAX_AUTH_RSA)) {
 			if (ast_strlen_zero(secret)) {
-				ast_log(LOG_NOTICE, "Asked to authenticate to %s with an RSA key, but they don't allow RSA authentication\n", ast_sockaddr_stringify_addr(addr));
+				ast_log(LOG_WARNING, "Asked to authenticate to %s with an RSA key, but they don't allow RSA authentication\n", ast_sockaddr_stringify_addr(addr));
 			}
 		} else if (ast_strlen_zero(challenge)) {
-			ast_log(LOG_NOTICE, "No challenge provided for RSA authentication to %s\n", ast_sockaddr_stringify_addr(addr));
+			ast_log(LOG_WARNING, "No challenge provided for RSA authentication to %s\n", ast_sockaddr_stringify_addr(addr));
 		} else {
 			char sig[256];
 			struct ast_key *key;
 			key = ast_key_get(keyn, AST_KEY_PRIVATE);
 			if (!key) {
-				ast_log(LOG_NOTICE, "Unable to find private key '%s'\n", keyn);
+				ast_log(LOG_WARNING, "Unable to find private key '%s'\n", keyn);
 			} else {
 				if (ast_sign(key, (char*)challenge, sig)) {
-					ast_log(LOG_NOTICE, "Unable to sign challenge with key\n");
+					ast_log(LOG_WARNING, "Unable to sign challenge with key\n");
 					res = -1;
 				} else {
 					iax_ie_append_str(ied, IAX_IE_RSA_RESULT, sig);
+					if (pvt) {
+						pvt->eff_auth_method = IAX_AUTH_RSA;
+					}
 					res = 0;
 				}
 			}
@@ -8465,11 +8523,15 @@ static int authenticate(const char *challenge, const char *secret, const char *k
 				sprintf(digres + (x << 1),  "%02hhx", digest[x]); /* safe */
 			if (pvt) {
 				build_encryption_keys(digest, pvt);
+				pvt->eff_auth_method = IAX_AUTH_MD5;
 			}
 			iax_ie_append_str(ied, IAX_IE_MD5_RESULT, digres);
 			res = 0;
 		} else if (authmethods & IAX_AUTH_PLAINTEXT) {
 			iax_ie_append_str(ied, IAX_IE_PASSWORD, secret);
+			if (pvt) {
+				pvt->eff_auth_method = IAX_AUTH_PLAINTEXT;
+			}
 			res = 0;
 		} else
 			ast_log(LOG_WARNING, "No way to send secret to peer '%s' (their methods: %d)\n", ast_sockaddr_stringify_addr(addr), authmethods);
@@ -11311,7 +11373,7 @@ static int socket_process_helper(struct iax2_thread *thread)
 				}
 				if (authenticate_verify(iaxs[fr->callno], &ies)) {
 					if (authdebug)
-						ast_log(LOG_NOTICE, "Host %s failed to authenticate as %s\n", ast_sockaddr_stringify(&addr),
+						ast_log(LOG_WARNING, "Host %s failed to authenticate as %s\n", ast_sockaddr_stringify(&addr),
 								iaxs[fr->callno]->username);
 					memset(&ied0, 0, sizeof(ied0));
 					auth_fail(fr->callno, IAX_COMMAND_REJECT);
@@ -11324,7 +11386,7 @@ static int socket_process_helper(struct iax2_thread *thread)
 					exists = 0;
 				if (strcmp(iaxs[fr->callno]->exten, "TBD") && !exists) {
 					if (authdebug)
-						ast_log(LOG_NOTICE, "Rejected connect attempt from %s, request '%s@%s' does not exist\n",
+						ast_log(LOG_WARNING, "Rejected connect attempt from %s, request '%s@%s' does not exist\n",
 								ast_sockaddr_stringify(&addr),
 								iaxs[fr->callno]->exten,
 								iaxs[fr->callno]->context);
@@ -11379,12 +11441,12 @@ static int socket_process_helper(struct iax2_thread *thread)
 						if (!format) {
 							if (authdebug) {
 								if (ast_test_flag64(iaxs[fr->callno], IAX_CODEC_NOCAP)) {
-									ast_log(LOG_NOTICE, "Rejected connect attempt from %s, requested '%s' incompatible with our capability '%s'.\n",
+									ast_log(LOG_WARNING, "Rejected connect attempt from %s, requested '%s' incompatible with our capability '%s'.\n",
 											ast_sockaddr_stringify(&addr),
 										iax2_getformatname_multiple(iaxs[fr->callno]->peerformat, &peer_form_buf),
 										iax2_getformatname_multiple(iaxs[fr->callno]->capability, &cap_buf));
 								} else {
-									ast_log(LOG_NOTICE, "Rejected connect attempt from %s, requested/capability '%s'/'%s' incompatible with our capability '%s'.\n",
+									ast_log(LOG_WARNING, "Rejected connect attempt from %s, requested/capability '%s'/'%s' incompatible with our capability '%s'.\n",
 										ast_sockaddr_stringify(&addr),
 										iax2_getformatname_multiple(iaxs[fr->callno]->peerformat, &peer_form_buf),
 										iax2_getformatname_multiple(iaxs[fr->callno]->peercapability, &peer_buf),
@@ -11437,12 +11499,12 @@ static int socket_process_helper(struct iax2_thread *thread)
 									iax2_getformatname_multiple(iaxs[fr->callno]->peercapability & iaxs[fr->callno]->capability, &cap_buf));
 								if (authdebug) {
 									if (ast_test_flag64(iaxs[fr->callno], IAX_CODEC_NOCAP)) {
-										ast_log(LOG_NOTICE, "Rejected connect attempt from %s, requested '%s' incompatible with our capability '%s'.\n",
+										ast_log(LOG_WARNING, "Rejected connect attempt from %s, requested '%s' incompatible with our capability '%s'.\n",
 											ast_sockaddr_stringify(&addr),
 											iax2_getformatname_multiple(iaxs[fr->callno]->peerformat, &peer_form_buf),
 											iax2_getformatname_multiple(iaxs[fr->callno]->capability, &cap_buf));
 									} else {
-										ast_log(LOG_NOTICE, "Rejected connect attempt from %s, requested/capability '%s'/'%s' incompatible with our capability '%s'.\n",
+										ast_log(LOG_WARNING, "Rejected connect attempt from %s, requested/capability '%s'/'%s' incompatible with our capability '%s'.\n",
 											ast_sockaddr_stringify(&addr),
 											iax2_getformatname_multiple(iaxs[fr->callno]->peerformat, &peer_form_buf),
 											iax2_getformatname_multiple(iaxs[fr->callno]->peercapability, &peer_buf),
@@ -11466,8 +11528,12 @@ static int socket_process_helper(struct iax2_thread *thread)
 						iax_ie_append_versioned_uint64(&ied1, IAX_IE_FORMAT2, 0, format);
 						send_command(iaxs[fr->callno], AST_FRAME_IAX, IAX_COMMAND_ACCEPT, 0, ied1.buf, ied1.pos, -1);
 						if (strcmp(iaxs[fr->callno]->exten, "TBD")) {
+							char authmethodnames[AUTH_METHOD_NAMES_BUFSIZE];
 							ast_set_flag(&iaxs[fr->callno]->state, IAX_STATE_STARTED);
 							ast_verb(3, "Accepting AUTHENTICATED call from %s:\n"
+											"%srequested auth methods = (%s),\n"
+											"%sactual auth method = %s,\n"
+											"%sencrypted = %s,\n"
 											"%srequested format = %s,\n"
 											"%srequested prefs = %s,\n"
 											"%sactual format = %s,\n"
@@ -11475,6 +11541,12 @@ static int socket_process_helper(struct iax2_thread *thread)
 											"%spriority = %s\n",
 											ast_sockaddr_stringify(&addr),
 											VERBOSE_PREFIX_4,
+											auth_method_names(iaxs[fr->callno]->authmethods, authmethodnames),
+											VERBOSE_PREFIX_4,
+											auth_method_labels[iaxs[fr->callno]->eff_auth_method],
+											VERBOSE_PREFIX_4,
+											IAX_CALLENCRYPTED(iaxs[fr->callno]) ? "yes" : "no",
+											VERBOSE_PREFIX_4,
 											iax2_getformatname(iaxs[fr->callno]->peerformat),
 											VERBOSE_PREFIX_4,
 											caller_pref_buf,
@@ -11543,7 +11615,7 @@ immediatedial:
 					ast_string_field_set(iaxs[fr->callno], exten, ies.called_number ? ies.called_number : "s");
 					if (!ast_exists_extension(NULL, iaxs[fr->callno]->context, iaxs[fr->callno]->exten, 1, iaxs[fr->callno]->cid_num)) {
 						if (authdebug)
-							ast_log(LOG_NOTICE, "Rejected dial attempt from %s, request '%s@%s' does not exist\n",
+							ast_log(LOG_WARNING, "Rejected dial attempt from %s, request '%s@%s' does not exist\n",
 									ast_sockaddr_stringify(&addr),
 									iaxs[fr->callno]->exten,
 									iaxs[fr->callno]->context);

channels/chan_pjsip.c

@@ -3266,6 +3266,8 @@ static struct ast_custom_function session_refresh_function = {
 	.write = pjsip_acf_session_refresh_write,
 };
 
+static char *app_pjsip_hangup = "PJSIPHangup";
+
 /*!
  * \brief Load the module
  *
@@ -3323,6 +3325,13 @@ static int load_module(void)
 		goto end;
 	}
 
+	if (ast_register_application_xml(app_pjsip_hangup, pjsip_app_hangup)) {
+		ast_log(LOG_WARNING, "Unable to register PJSIPHangup dialplan application\n");
+		goto end;
+	}
+	ast_manager_register_xml(app_pjsip_hangup, EVENT_FLAG_SYSTEM | EVENT_FLAG_CALL, pjsip_action_hangup);
+
+
 	ast_sip_register_service(&refer_callback_module);
 
 	ast_sip_session_register_supplement(&chan_pjsip_supplement);
@@ -3370,6 +3379,9 @@ end:
 	ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
 	ast_custom_function_unregister(&chan_pjsip_parse_uri_function);
 	ast_custom_function_unregister(&session_refresh_function);
+	ast_unregister_application(app_pjsip_hangup);
+	ast_manager_unregister(app_pjsip_hangup);
+
 	ast_channel_unregister(&chan_pjsip_tech);
 	ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
 
@@ -3399,6 +3411,8 @@ static int unload_module(void)
 	ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
 	ast_custom_function_unregister(&chan_pjsip_parse_uri_function);
 	ast_custom_function_unregister(&session_refresh_function);
+	ast_unregister_application(app_pjsip_hangup);
+	ast_manager_unregister(app_pjsip_hangup);
 
 	ast_channel_unregister(&chan_pjsip_tech);
 	ao2_ref(chan_pjsip_tech.capabilities, -1);

channels/chan_rtp.c

@@ -129,7 +129,8 @@ static struct ast_format *derive_format_from_cap(struct ast_format_cap *cap)
 		 * assignments. Signed linear @ 8kHz does not map, so if that is our
 		 * only capability, we force μ-law instead.
 		 */
-		fmt = ast_format_ulaw;
+		ao2_ref(fmt, -1);
+		fmt = ao2_bump(ast_format_ulaw);
 	}
 
 	return fmt;
@@ -211,7 +212,7 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
 	}
 
 	chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids,
-		requestor, 0, "MulticastRTP/%p", instance);
+		requestor, 0, "MulticastRTP/%s-%p", args.destination, instance);
 	if (!chan) {
 		ast_rtp_instance_destroy(instance);
 		goto failure;
@@ -249,6 +250,7 @@ failure:
 enum {
 	OPT_RTP_CODEC =  (1 << 0),
 	OPT_RTP_ENGINE = (1 << 1),
+	OPT_RTP_GLUE = (1 << 2),
 };
 
 enum {
@@ -263,8 +265,14 @@ AST_APP_OPTIONS(unicast_rtp_options, BEGIN_OPTIONS
 	AST_APP_OPTION_ARG('c', OPT_RTP_CODEC, OPT_ARG_RTP_CODEC),
 	/*! Set the RTP engine to use for unicast RTP */
 	AST_APP_OPTION_ARG('e', OPT_RTP_ENGINE, OPT_ARG_RTP_ENGINE),
+	/*! Provide RTP glue for the channel */
+	AST_APP_OPTION('g', OPT_RTP_GLUE),
 END_OPTIONS );
 
+static const struct ast_datastore_info chan_rtp_datastore_info = {
+	.type = "CHAN_RTP_GLUE",
+};
+
 /*! \brief Function called when we should prepare to call the unicast destination */
 static struct ast_channel *unicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
 {
@@ -372,6 +380,13 @@ static struct ast_channel *unicast_rtp_request(const char *type, struct ast_form
 
 	ast_channel_tech_set(chan, &unicast_rtp_tech);
 
+	if (ast_test_flag(&opts, OPT_RTP_GLUE)) {
+		struct ast_datastore *datastore;
+		if ((datastore = ast_datastore_alloc(&chan_rtp_datastore_info, NULL))) {
+			ast_channel_datastore_add(chan, datastore);
+		}
+	}
+
 	ast_format_cap_append(caps, fmt, 0);
 	ast_channel_nativeformats_set(chan, caps);
 	ast_channel_set_writeformat(chan, fmt);
@@ -401,6 +416,61 @@ failure:
 	return NULL;
 }
 
+/*! \brief Function called by RTP engine to get peer capabilities */
+static void chan_rtp_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
+{
+	SCOPE_ENTER(1, "%s Native formats %s\n", ast_channel_name(chan),
+		ast_str_tmp(AST_FORMAT_CAP_NAMES_LEN, ast_format_cap_get_names(ast_channel_nativeformats(chan), &STR_TMP)));
+	ast_format_cap_append_from_cap(result, ast_channel_nativeformats(chan), AST_MEDIA_TYPE_UNKNOWN);
+	SCOPE_EXIT_RTN();
+}
+
+/*! \brief Function called by RTP engine to change where the remote party should send media.
+ *
+ * chan_rtp is not able to actually update the peer, so this function has no effect.
+ * */
+static int chan_rtp_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *tpeer, const struct ast_format_cap *cap, int nat_active)
+{
+	return -1;
+}
+
+/*! \brief Function called by RTP engine to get local audio RTP peer */
+static enum ast_rtp_glue_result chan_rtp_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
+{
+	return AST_RTP_GLUE_RESULT_FORBID;
+}
+
+/*! \brief Function called by RTP engine to get local audio RTP peer */
+static enum ast_rtp_glue_result chan_rtp_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
+{
+	struct ast_rtp_instance *rtp_instance = ast_channel_tech_pvt(chan);
+	struct ast_datastore *datastore;
+
+	if (!rtp_instance) {
+		return AST_RTP_GLUE_RESULT_FORBID;
+	}
+
+	if ((datastore = ast_channel_datastore_find(chan, &chan_rtp_datastore_info, NULL))) {
+		ao2_ref(datastore, -1);
+
+		*instance = rtp_instance;
+		ao2_ref(*instance, +1);
+
+		return AST_RTP_GLUE_RESULT_LOCAL;
+	}
+
+	return AST_RTP_GLUE_RESULT_FORBID;
+}
+
+/*! \brief Local glue for interacting with the RTP engine core */
+static struct ast_rtp_glue unicast_rtp_glue = {
+	.type = "UnicastRTP",
+	.get_rtp_info = chan_rtp_get_rtp_peer,
+	.get_vrtp_info = chan_rtp_get_vrtp_peer,
+	.get_codec = chan_rtp_get_codec,
+	.update_peer = chan_rtp_set_rtp_peer,
+};
+
 /*! \brief Function called when our module is unloaded */
 static int unload_module(void)
 {
@@ -412,6 +482,8 @@ static int unload_module(void)
 	ao2_cleanup(unicast_rtp_tech.capabilities);
 	unicast_rtp_tech.capabilities = NULL;
 
+	ast_rtp_glue_unregister(&unicast_rtp_glue);
+
 	return 0;
 }
 
@@ -421,6 +493,9 @@ static int load_module(void)
 	if (!(multicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
 		return AST_MODULE_LOAD_DECLINE;
 	}
+
+	ast_rtp_glue_register(&unicast_rtp_glue);
+
 	ast_format_cap_append_by_type(multicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
 	if (ast_channel_register(&multicast_rtp_tech)) {
 		ast_log(LOG_ERROR, "Unable to register channel class 'MulticastRTP'\n");

channels/iax2/parser.c

@@ -424,7 +424,7 @@ static void dump_ies(unsigned char *iedata, int len)
 
 	if (len < 2)
 		return;
-	while(len > 2) {
+	while(len >= 2) {
 		ie = iedata[0];
 		ielen = iedata[1];
 		if (ielen + 2> len) {
@@ -1063,7 +1063,7 @@ int iax_parse_ies(struct iax_ies *ies, unsigned char *data, int datalen)
 			if (len == (int)sizeof(unsigned int)) {
 				ies->calling_ani2 = ntohl(get_unaligned_uint32(data + 2));
 			} else {
-				snprintf(tmp, (int)sizeof(tmp), "callingani2 was %d long: %s\n", len, data + 2);
+				snprintf(tmp, sizeof(tmp), "Expected callingani2 to be %zu bytes but was %d\n", sizeof(unsigned int), len);
 				errorf(tmp);
 			}
 			break;

channels/pjsip/cli_commands.c

@@ -343,6 +343,7 @@ static int cli_channelstats_print_body(void *obj, void *arg, int flags)
 	struct ast_sip_session *session;
 	struct ast_sip_session_media *media;
 	struct ast_rtp_instance_stats stats;
+	struct ast_stream *stream;
 	char *print_name = NULL;
 	char *print_time = alloca(32);
 	char codec_in_use[7];
@@ -359,16 +360,29 @@ static int cli_channelstats_print_body(void *obj, void *arg, int flags)
 
 	cpvt = ast_channel_tech_pvt(channel);
 	session = cpvt ? cpvt->session : NULL;
-	if (!session) {
+
+	if (!session
+		|| !session->active_media_state
+		|| !session->active_media_state->topology) {
 		ast_str_append(&context->output_buffer, 0, " %s not valid\n", snapshot->base->name);
 		ast_channel_unlock(channel);
 		ao2_cleanup(channel);
 		return 0;
 	}
 
+	stream = ast_stream_topology_get_first_stream_by_type(
+		session->active_media_state->topology, AST_MEDIA_TYPE_AUDIO);
+
+	if (!stream) {
+		ast_str_append(&context->output_buffer, 0, " %s no audio streams\n", snapshot->base->name);
+		ast_channel_unlock(channel);
+		ao2_cleanup(channel);
+		return 0;
+	}
+
 	media = session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
-	if (!media || !media->rtp) {
-		ast_str_append(&context->output_buffer, 0, " %s not valid\n", snapshot->base->name);
+	if (!media || media->type != AST_MEDIA_TYPE_AUDIO || !media->rtp) {
+		ast_str_append(&context->output_buffer, 0, " %s corrupted default audio session\n", snapshot->base->name);
 		ast_channel_unlock(channel);
 		ao2_cleanup(channel);
 		return 0;

channels/pjsip/dialplan_functions.c

@@ -29,563 +29,6 @@
 	<support_level>core</support_level>
  ***/
 
-/*** DOCUMENTATION
-<function name="PJSIP_DIAL_CONTACTS" language="en_US">
-	<synopsis>
-		Return a dial string for dialing all contacts on an AOR.
-	</synopsis>
-	<syntax>
-		<parameter name="endpoint" required="true">
-			<para>Name of the endpoint</para>
-		</parameter>
-		<parameter name="aor" required="false">
-			<para>Name of an AOR to use, if not specified the configured AORs on the endpoint are used</para>
-		</parameter>
-		<parameter name="request_user" required="false">
-			<para>Optional request user to use in the request URI</para>
-		</parameter>
-	</syntax>
-	<description>
-		<para>Returns a properly formatted dial string for dialing all contacts on an AOR.</para>
-	</description>
-</function>
-<function name="PJSIP_MEDIA_OFFER" language="en_US">
-	<synopsis>
-		Media and codec offerings to be set on an outbound SIP channel prior to dialing.
-	</synopsis>
-	<syntax>
-		<parameter name="media" required="true">
-			<para>types of media offered</para>
-		</parameter>
-	</syntax>
-	<description>
-		<para>When read, returns the codecs offered based upon the media choice.</para>
-		<para>When written, sets the codecs to offer when an outbound dial attempt is made,
-		or when a session refresh is sent using <replaceable>PJSIP_SEND_SESSION_REFRESH</replaceable>.
-		</para>
-	</description>
-	<see-also>
-		<ref type="function">PJSIP_SEND_SESSION_REFRESH</ref>
-	</see-also>
-</function>
-<function name="PJSIP_DTMF_MODE" language="en_US">
-	<since>
-		<version>13.18.0</version>
-		<version>14.7.0</version>
-		<version>15.1.0</version>
-		<version>16.0.0</version>
-	</since>
-	<synopsis>
-		Get or change the DTMF mode for a SIP call.
-	</synopsis>
-	<syntax>
-	</syntax>
-	<description>
-		<para>When read, returns the current DTMF mode</para>
-		<para>When written, sets the current DTMF mode</para>
-		<para>This function uses the same DTMF mode naming as the dtmf_mode configuration option</para>
-	</description>
-</function>
-<function name="PJSIP_MOH_PASSTHROUGH" language="en_US">
-	<synopsis>
-		Get or change the on-hold behavior for a SIP call.
-	</synopsis>
-	<syntax>
-	</syntax>
-	<description>
-		<para>When read, returns the current moh passthrough mode</para>
-		<para>When written, sets the current moh passthrough mode</para>
-		<para>If <replaceable>yes</replaceable>, on-hold re-INVITEs are sent. If <replaceable>no</replaceable>, music on hold is generated.</para>
-		<para>This function can be used to override the moh_passthrough configuration option</para>
-	</description>
-</function>
-<function name="PJSIP_SEND_SESSION_REFRESH" language="en_US">
-	<since>
-		<version>13.12.0</version>
-		<version>14.1.0</version>
-		<version>15.0.0</version>
-	</since>
-	<synopsis>
-		W/O: Initiate a session refresh via an UPDATE or re-INVITE on an established media session
-	</synopsis>
-	<syntax>
-		<parameter name="update_type" required="false">
-			<para>The type of update to send. Default is <literal>invite</literal>.</para>
-			<enumlist>
-				<enum name="invite">
-					<para>Send the session refresh as a re-INVITE.</para>
-				</enum>
-				<enum name="update">
-					<para>Send the session refresh as an UPDATE.</para>
-				</enum>
-			</enumlist>
-		</parameter>
-	</syntax>
-	<description>
-		<para>This function will cause the PJSIP stack to immediately refresh
-		the media session for the channel. This will be done using either a
-		re-INVITE (default) or an UPDATE request.
-		</para>
-		<para>This is most useful when combined with the <replaceable>PJSIP_MEDIA_OFFER</replaceable>
-		dialplan function, as it allows the formats in use on a channel to be
-		re-negotiated after call setup.</para>
-		<warning>
-			<para>The formats the endpoint supports are <emphasis>not</emphasis>
-			checked or enforced by this function. Using this function to offer
-			formats not supported by the endpoint <emphasis>may</emphasis> result
-			in a loss of media.</para>
-		</warning>
-		<example title="Re-negotiate format to g722">
-		 ; Within some existing extension on an answered channel
-		 same => n,Set(PJSIP_MEDIA_OFFER(audio)=!all,g722)
-		 same => n,Set(PJSIP_SEND_SESSION_REFRESH()=invite)
-		</example>
-	</description>
-	<see-also>
-		<ref type="function">PJSIP_MEDIA_OFFER</ref>
-	</see-also>
-</function>
-<function name="PJSIP_PARSE_URI" language="en_US">
-	<since>
-		<version>13.24.0</version>
-		<version>16.1.0</version>
-		<version>17.0.0</version>
-	</since>
-	<synopsis>
-		Parse an uri and return a type part of the URI.
-	</synopsis>
-	<syntax>
-		<parameter name="uri" required="true">
-			<para>URI to parse</para>
-		</parameter>
-		<parameter name="type" required="true">
-			<para>The <literal>type</literal> parameter specifies which URI part to read</para>
-			<enumlist>
-				<enum name="display">
-					<para>Display name.</para>
-				</enum>
-				<enum name="scheme">
-					<para>URI scheme.</para>
-				</enum>
-				<enum name="user">
-					<para>User part.</para>
-				</enum>
-				<enum name="passwd">
-					<para>Password part.</para>
-				</enum>
-				<enum name="host">
-					<para>Host part.</para>
-				</enum>
-				<enum name="port">
-					<para>Port number, or zero.</para>
-				</enum>
-				<enum name="user_param">
-					<para>User parameter.</para>
-				</enum>
-				<enum name="method_param">
-					<para>Method parameter.</para>
-				</enum>
-				<enum name="transport_param">
-					<para>Transport parameter.</para>
-				</enum>
-				<enum name="ttl_param">
-					<para>TTL param, or -1.</para>
-				</enum>
-				<enum name="lr_param">
-					<para>Loose routing param, or zero.</para>
-				</enum>
-				<enum name="maddr_param">
-					<para>Maddr param.</para>
-				</enum>
-			</enumlist>
-		</parameter>
-	</syntax>
-	<description>
-		<para>Parse an URI and return a specified part of the URI.</para>
-	</description>
-</function>
-<info name="CHANNEL" language="en_US" tech="PJSIP">
-	<enumlist>
-		<enum name="rtp">
-			<para>R/O Retrieve media related information.</para>
-			<parameter name="type" required="true">
-				<para>When <replaceable>rtp</replaceable> is specified, the
-				<literal>type</literal> parameter must be provided. It specifies
-				which RTP parameter to read.</para>
-				<enumlist>
-					<enum name="src">
-						<para>Retrieve the local address for RTP.</para>
-					</enum>
-					<enum name="dest">
-						<para>Retrieve the remote address for RTP.</para>
-					</enum>
-					<enum name="direct">
-						<para>If direct media is enabled, this address is the remote address
-						used for RTP.</para>
-					</enum>
-					<enum name="secure">
-						<para>Whether or not the media stream is encrypted.</para>
-						<enumlist>
-							<enum name="0">
-								<para>The media stream is not encrypted.</para>
-							</enum>
-							<enum name="1">
-								<para>The media stream is encrypted.</para>
-							</enum>
-						</enumlist>
-					</enum>
-					<enum name="hold">
-						<para>Whether or not the media stream is currently restricted
-						due to a call hold.</para>
-						<enumlist>
-							<enum name="0">
-								<para>The media stream is not held.</para>
-							</enum>
-							<enum name="1">
-								<para>The media stream is held.</para>
-							</enum>
-						</enumlist>
-					</enum>
-				</enumlist>
-			</parameter>
-			<parameter name="media_type" required="false">
-				<para>When <replaceable>rtp</replaceable> is specified, the
-				<literal>media_type</literal> parameter may be provided. It specifies
-				which media stream the chosen RTP parameter should be retrieved
-				from.</para>
-				<enumlist>
-					<enum name="audio">
-						<para>Retrieve information from the audio media stream.</para>
-						<note><para>If not specified, <literal>audio</literal> is used
-						by default.</para></note>
-					</enum>
-					<enum name="video">
-						<para>Retrieve information from the video media stream.</para>
-					</enum>
-				</enumlist>
-			</parameter>
-		</enum>
-		<enum name="rtcp">
-			<para>R/O Retrieve RTCP statistics.</para>
-			<parameter name="statistic" required="true">
-				<para>When <replaceable>rtcp</replaceable> is specified, the
-				<literal>statistic</literal> parameter must be provided. It specifies
-				which RTCP statistic parameter to read.</para>
-				<enumlist>
-					<enum name="all">
-						<para>Retrieve a summary of all RTCP statistics.</para>
-						<para>The following data items are returned in a semi-colon
-						delineated list:</para>
-						<enumlist>
-							<enum name="ssrc">
-								<para>Our Synchronization Source identifier</para>
-							</enum>
-							<enum name="themssrc">
-								<para>Their Synchronization Source identifier</para>
-							</enum>
-							<enum name="lp">
-								<para>Our lost packet count</para>
-							</enum>
-							<enum name="rxjitter">
-								<para>Received packet jitter</para>
-							</enum>
-							<enum name="rxcount">
-								<para>Received packet count</para>
-							</enum>
-							<enum name="txjitter">
-								<para>Transmitted packet jitter</para>
-							</enum>
-							<enum name="txcount">
-								<para>Transmitted packet count</para>
-							</enum>
-							<enum name="rlp">
-								<para>Remote lost packet count</para>
-							</enum>
-							<enum name="rtt">
-								<para>Round trip time</para>
-							</enum>
-							<enum name="txmes">
-								<para>Transmitted Media Experience Score</para>
-							</enum>
-							<enum name="rxmes">
-								<para>Received Media Experience Score</para>
-							</enum>
-						</enumlist>
-					</enum>
-					<enum name="all_jitter">
-						<para>Retrieve a summary of all RTCP Jitter statistics.</para>
-						<para>The following data items are returned in a semi-colon
-						delineated list:</para>
-						<enumlist>
-							<enum name="minrxjitter">
-								<para>Our minimum jitter</para>
-							</enum>
-							<enum name="maxrxjitter">
-								<para>Our max jitter</para>
-							</enum>
-							<enum name="avgrxjitter">
-								<para>Our average jitter</para>
-							</enum>
-							<enum name="stdevrxjitter">
-								<para>Our jitter standard deviation</para>
-							</enum>
-							<enum name="reported_minjitter">
-								<para>Their minimum jitter</para>
-							</enum>
-							<enum name="reported_maxjitter">
-								<para>Their max jitter</para>
-							</enum>
-							<enum name="reported_avgjitter">
-								<para>Their average jitter</para>
-							</enum>
-							<enum name="reported_stdevjitter">
-								<para>Their jitter standard deviation</para>
-							</enum>
-						</enumlist>
-					</enum>
-					<enum name="all_loss">
-						<para>Retrieve a summary of all RTCP packet loss statistics.</para>
-						<para>The following data items are returned in a semi-colon
-						delineated list:</para>
-						<enumlist>
-							<enum name="minrxlost">
-								<para>Our minimum lost packets</para>
-							</enum>
-							<enum name="maxrxlost">
-								<para>Our max lost packets</para>
-							</enum>
-							<enum name="avgrxlost">
-								<para>Our average lost packets</para>
-							</enum>
-							<enum name="stdevrxlost">
-								<para>Our lost packets standard deviation</para>
-							</enum>
-							<enum name="reported_minlost">
-								<para>Their minimum lost packets</para>
-							</enum>
-							<enum name="reported_maxlost">
-								<para>Their max lost packets</para>
-							</enum>
-							<enum name="reported_avglost">
-								<para>Their average lost packets</para>
-							</enum>
-							<enum name="reported_stdevlost">
-								<para>Their lost packets standard deviation</para>
-							</enum>
-						</enumlist>
-					</enum>
-					<enum name="all_rtt">
-						<para>Retrieve a summary of all RTCP round trip time information.</para>
-						<para>The following data items are returned in a semi-colon
-						delineated list:</para>
-						<enumlist>
-							<enum name="minrtt">
-								<para>Minimum round trip time</para>
-							</enum>
-							<enum name="maxrtt">
-								<para>Maximum round trip time</para>
-							</enum>
-							<enum name="avgrtt">
-								<para>Average round trip time</para>
-							</enum>
-							<enum name="stdevrtt">
-								<para>Standard deviation round trip time</para>
-							</enum>
-						</enumlist>
-					</enum>
-					<enum name="all_mes">
-						<para>Retrieve a summary of all RTCP Media Experience Score information.</para>
-						<para>The following data items are returned in a semi-colon
-						delineated list:</para>
-						<enumlist>
-							<enum name="minmes">
-								<para>Minimum MES based on us analysing received packets.</para>
-							</enum>
-							<enum name="maxmes">
-								<para>Maximum MES based on us analysing received packets.</para>
-							</enum>
-							<enum name="avgmes">
-								<para>Average MES based on us analysing received packets.</para>
-							</enum>
-							<enum name="stdevmes">
-								<para>Standard deviation MES based on us analysing received packets.</para>
-							</enum>
-							<enum name="reported_minmes">
-								<para>Minimum MES based on data we get in Sender and Receiver Reports sent by the remote end</para>
-							</enum>
-							<enum name="reported_maxmes">
-								<para>Maximum MES based on data we get in Sender and Receiver Reports sent by the remote end</para>
-							</enum>
-							<enum name="reported_avgmes">
-								<para>Average MES based on data we get in Sender and Receiver Reports sent by the remote end</para>
-							</enum>
-							<enum name="reported_stdevmes">
-								<para>Standard deviation MES based on data we get in Sender and Receiver Reports sent by the remote end</para>
-							</enum>
-						</enumlist>
-					</enum>
-					<enum name="txcount"><para>Transmitted packet count</para></enum>
-					<enum name="rxcount"><para>Received packet count</para></enum>
-					<enum name="txjitter"><para>Transmitted packet jitter</para></enum>
-					<enum name="rxjitter"><para>Received packet jitter</para></enum>
-					<enum name="remote_maxjitter"><para>Their max jitter</para></enum>
-					<enum name="remote_minjitter"><para>Their minimum jitter</para></enum>
-					<enum name="remote_normdevjitter"><para>Their average jitter</para></enum>
-					<enum name="remote_stdevjitter"><para>Their jitter standard deviation</para></enum>
-					<enum name="local_maxjitter"><para>Our max jitter</para></enum>
-					<enum name="local_minjitter"><para>Our minimum jitter</para></enum>
-					<enum name="local_normdevjitter"><para>Our average jitter</para></enum>
-					<enum name="local_stdevjitter"><para>Our jitter standard deviation</para></enum>
-					<enum name="txploss"><para>Transmitted packet loss</para></enum>
-					<enum name="rxploss"><para>Received packet loss</para></enum>
-					<enum name="remote_maxrxploss"><para>Their max lost packets</para></enum>
-					<enum name="remote_minrxploss"><para>Their minimum lost packets</para></enum>
-					<enum name="remote_normdevrxploss"><para>Their average lost packets</para></enum>
-					<enum name="remote_stdevrxploss"><para>Their lost packets standard deviation</para></enum>
-					<enum name="local_maxrxploss"><para>Our max lost packets</para></enum>
-					<enum name="local_minrxploss"><para>Our minimum lost packets</para></enum>
-					<enum name="local_normdevrxploss"><para>Our average lost packets</para></enum>
-					<enum name="local_stdevrxploss"><para>Our lost packets standard deviation</para></enum>
-					<enum name="rtt"><para>Round trip time</para></enum>
-					<enum name="maxrtt"><para>Maximum round trip time</para></enum>
-					<enum name="minrtt"><para>Minimum round trip time</para></enum>
-					<enum name="normdevrtt"><para>Average round trip time</para></enum>
-					<enum name="stdevrtt"><para>Standard deviation round trip time</para></enum>
-					<enum name="local_ssrc"><para>Our Synchronization Source identifier</para></enum>
-					<enum name="remote_ssrc"><para>Their Synchronization Source identifier</para></enum>
-					<enum name="txmes"><para>
-					Current MES based on us analyzing rtt, jitter and loss
-					in the actual received RTP stream received from the remote end.
-					I.E.  This is the MES for the incoming audio stream.
-					</para></enum>
-					<enum name="rxmes"><para>
-					Current MES based on rtt and the jitter and loss values in
-					RTCP sender and receiver reports we receive from the
-					remote end. I.E.  This is the MES for the outgoing audio stream.
-					</para></enum>
-					<enum name="remote_maxmes"><para>Max MES based on data we get in Sender and Receiver Reports sent by the remote end</para></enum>
-					<enum name="remote_minmes"><para>Min MES based on data we get in Sender and Receiver Reports sent by the remote end</para></enum>
-					<enum name="remote_normdevmes"><para>Average MES based on data we get in Sender and Receiver Reports sent by the remote end</para></enum>
-					<enum name="remote_stdevmes"><para>Standard deviation MES based on data we get in Sender and Receiver Reports sent by the remote end</para></enum>
-					<enum name="local_maxmes"><para>Max MES based on us analyzing the received RTP stream</para></enum>
-					<enum name="local_minmes"><para>Min MES based on us analyzing the received RTP stream</para></enum>
-					<enum name="local_normdevmes"><para>Average MES based on us analyzing the received RTP stream</para></enum>
-					<enum name="local_stdevmes"><para>Standard deviation MES based on us analyzing the received RTP stream</para></enum>
-				</enumlist>
-			</parameter>
-			<parameter name="media_type" required="false">
-				<para>When <replaceable>rtcp</replaceable> is specified, the
-				<literal>media_type</literal> parameter may be provided. It specifies
-				which media stream the chosen RTCP parameter should be retrieved
-				from.</para>
-				<enumlist>
-					<enum name="audio">
-						<para>Retrieve information from the audio media stream.</para>
-						<note><para>If not specified, <literal>audio</literal> is used
-						by default.</para></note>
-					</enum>
-					<enum name="video">
-						<para>Retrieve information from the video media stream.</para>
-					</enum>
-				</enumlist>
-			</parameter>
-		</enum>
-		<enum name="endpoint">
-			<para>R/O The name of the endpoint associated with this channel.
-			Use the <replaceable>PJSIP_ENDPOINT</replaceable> function to obtain
-			further endpoint related information.</para>
-		</enum>
-		<enum name="contact">
-			<para>R/O The name of the contact associated with this channel.
-			Use the <replaceable>PJSIP_CONTACT</replaceable> function to obtain
-			further contact related information. Note this may not be present and if so
-			is only available on outgoing legs.</para>
-		</enum>
-		<enum name="aor">
-			<para>R/O The name of the AOR associated with this channel.
-			Use the <replaceable>PJSIP_AOR</replaceable> function to obtain
-			further AOR related information. Note this may not be present and if so
-			is only available on outgoing legs.</para>
-		</enum>
-		<enum name="pjsip">
-			<para>R/O Obtain information about the current PJSIP channel and its
-			session.</para>
-			<parameter name="type" required="true">
-				<para>When <replaceable>pjsip</replaceable> is specified, the
-				<literal>type</literal> parameter must be provided. It specifies
-				which signalling parameter to read.</para>
-				<enumlist>
-					<enum name="call-id">
-						<para>The SIP call-id.</para>
-					</enum>
-					<enum name="secure">
-						<para>Whether or not the signalling uses a secure transport.</para>
-						<enumlist>
-							<enum name="0"><para>The signalling uses a non-secure transport.</para></enum>
-							<enum name="1"><para>The signalling uses a secure transport.</para></enum>
-						</enumlist>
-					</enum>
-					<enum name="target_uri">
-						<para>The contact URI where requests are sent.</para>
-					</enum>
-					<enum name="local_uri">
-						<para>The local URI.</para>
-					</enum>
-					<enum name="local_tag">
-						<para>Tag in From header</para>
-					</enum>
-					<enum name="remote_uri">
-						<para>The remote URI.</para>
-					</enum>
-					<enum name="remote_tag">
-						<para>Tag in To header</para>
-					</enum>
-					<enum name="request_uri">
-						<para>The request URI of the incoming <literal>INVITE</literal>
-						associated with the creation of this channel.</para>
-					</enum>
-					<enum name="t38state">
-						<para>The current state of any T.38 fax on this channel.</para>
-						<enumlist>
-							<enum name="DISABLED"><para>T.38 faxing is disabled on this channel.</para></enum>
-							<enum name="LOCAL_REINVITE"><para>Asterisk has sent a <literal>re-INVITE</literal> to the remote end to initiate a T.38 fax.</para></enum>
-							<enum name="REMOTE_REINVITE"><para>The remote end has sent a <literal>re-INVITE</literal> to Asterisk to initiate a T.38 fax.</para></enum>
-							<enum name="ENABLED"><para>A T.38 fax session has been enabled.</para></enum>
-							<enum name="REJECTED"><para>A T.38 fax session was attempted but was rejected.</para></enum>
-						</enumlist>
-					</enum>
-					<enum name="local_addr">
-						<para>On inbound calls, the full IP address and port number that
-						the <literal>INVITE</literal> request was received on. On outbound
-						calls, the full IP address and port number that the <literal>INVITE</literal>
-						request was transmitted from.</para>
-					</enum>
-					<enum name="remote_addr">
-						<para>On inbound calls, the full IP address and port number that
-						the <literal>INVITE</literal> request was received from. On outbound
-						calls, the full IP address and port number that the <literal>INVITE</literal>
-						request was transmitted to.</para>
-					</enum>
-				</enumlist>
-			</parameter>
-		</enum>
-	</enumlist>
-</info>
-<info name="CHANNEL_EXAMPLES" language="en_US" tech="PJSIP">
-	<example title="PJSIP specific CHANNEL examples">
-		; Log the current Call-ID
-		same => n,Log(NOTICE, ${CHANNEL(pjsip,call-id)})
-
-		; Log the destination address of the audio stream
-		same => n,Log(NOTICE, ${CHANNEL(rtp,dest)})
-
-		; Store the round-trip time associated with a
-		; video stream in the CDR field video-rtt
-		same => n,Set(CDR(video-rtt)=${CHANNEL(rtcp,rtt,video)})
-	</example>
-</info>
-***/
-
 #include "asterisk.h"
 
 #include <pjsip.h>
@@ -596,6 +39,7 @@
 #include "asterisk/module.h"
 #include "asterisk/acl.h"
 #include "asterisk/app.h"
+#include "asterisk/conversions.h"
 #include "asterisk/channel.h"
 #include "asterisk/stream.h"
 #include "asterisk/format.h"
@@ -1784,3 +1228,121 @@ int pjsip_acf_session_refresh_write(struct ast_channel *chan, const char *cmd, c
 
 	return ast_sip_push_task_wait_serializer(channel->session->serializer, refresh_write_cb, &rdata);
 }
+
+struct hangup_data {
+	struct ast_sip_session *session;
+	int response_code;
+};
+
+/*!
+ * \brief Serializer task to hangup channel
+ */
+static int pjsip_hangup(void *obj)
+{
+	struct hangup_data *hdata = obj;
+	pjsip_tx_data *packet = NULL;
+
+	if ((hdata->session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED) &&
+		(pjsip_inv_answer(hdata->session->inv_session, hdata->response_code, NULL, NULL, &packet) == PJ_SUCCESS)) {
+		ast_sip_session_send_response(hdata->session, packet);
+	}
+
+	return 0;
+}
+
+/*!
+ * \brief Callback that validates the response code
+ */
+static int response_code_validator(const char *channel_name,
+	const char *response) {
+	int response_code;
+
+	int rc = ast_str_to_int(response, &response_code);
+	if (rc != 0) {
+		response_code = ast_sip_str2rc(response);
+		if (response_code < 0) {
+			ast_log(LOG_WARNING, "%s: Unrecognized response code parameter '%s'."
+				" Defaulting to 603 DECLINE\n",
+				channel_name, response);
+			return PJSIP_SC_DECLINE;
+		}
+	}
+
+	if (response_code < 400 || response_code > 699) {
+		ast_log(LOG_WARNING, "%s: Response code %d is out of range 400 -> 699."
+			" Defaulting to 603 DECLINE\n",
+			channel_name, response_code);
+		return PJSIP_SC_DECLINE;
+	}
+	return response_code;
+}
+
+/*!
+ * \brief Called by pjsip_app_hangup and pjsip_action_hangup
+ *        to actually perform the hangup
+ */
+static void pjsip_app_hangup_handler(struct ast_channel *chan, int response_code)
+{
+	struct ast_sip_channel_pvt *channel;
+	struct hangup_data hdata = { NULL, -1 };
+	const char *tag = ast_channel_name(chan);
+
+	hdata.response_code = response_code;
+
+	ast_channel_lock(chan);
+	if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
+		ast_log(LOG_WARNING, "%s: Not a PJSIP channel\n", tag);
+		ast_channel_unlock(chan);
+		return;
+	}
+
+	channel = ast_channel_tech_pvt(chan);
+	hdata.session = channel->session;
+
+	if (hdata.session->inv_session->role != PJSIP_ROLE_UAS || (
+		hdata.session->inv_session->state != PJSIP_INV_STATE_INCOMING &&
+		hdata.session->inv_session->state != PJSIP_INV_STATE_EARLY)) {
+		ast_log(LOG_WARNING, "%s: Not an incoming channel or invalid state '%s'\n",
+			tag, pjsip_inv_state_name(hdata.session->inv_session->state));
+		ast_channel_unlock(chan);
+		return;
+	}
+
+	ast_channel_unlock(chan);
+
+	if (ast_sip_push_task_wait_serializer(channel->session->serializer,
+		pjsip_hangup, &hdata) != 0) {
+		ast_log(LOG_WARNING, "%s: failed to push hangup task to serializer\n", tag);
+	}
+
+	return;
+}
+
+/*!
+ * \brief PJSIPHangup Dialplan App
+ */
+int pjsip_app_hangup(struct ast_channel *chan, const char *data)
+{
+	int response_code;
+	const char *tag = ast_channel_name(chan);
+
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "%s: Missing response code parameter\n", tag);
+		return -1;
+	}
+
+	response_code = response_code_validator(tag, data);
+
+	pjsip_app_hangup_handler(chan, response_code);
+
+	return -1;
+}
+
+/*!
+ * \brief PJSIPHangup Manager Action
+ */
+int pjsip_action_hangup(struct mansession *s, const struct message *m)
+{
+	return ast_manager_hangup_helper(s, m,
+		pjsip_app_hangup_handler, response_code_validator);
+}

channels/pjsip/dialplan_functions_doc.xml

@@ -0,0 +1,659 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE docs SYSTEM "appdocsxml.dtd">
+<docs>
+	<application name="PJSIPHangup" language="en_US">
+		<synopsis>
+			Hangup an incoming PJSIP channel with a SIP response code
+		</synopsis>
+		<syntax>
+			<parameter name="Cause" required="true">
+				<para>May be one of...</para>
+				<enumlist>
+				<enum name="Response code"><para>A numeric response code in the range 400 ->699</para></enum>
+				<enum name="Response code name"><para>A response code name from
+				<literal>third-party/pjproject/source/pjsip/include/pjsip/sip_msg.h</literal>
+				such as <literal>USE_IDENTITY_HEADER</literal> or 
+				<literal>PJSIP_SC_USE_IDENTITY_HEADER</literal></para></enum>
+				</enumlist>
+			</parameter>
+		</syntax>
+		<description>
+			<para>
+			Hangs up an incoming PJSIP channel and returns the
+			specified SIP response code in the final response to the caller.
+			</para>
+			<para>
+			</para>
+			<warning><para>
+			This function must be called BEFORE anything that
+			might cause any other final (non 1XX) response to be sent.
+			For example calling <literal>Answer()</literal> or
+			<literal>Playback</literal> without the
+			<literal>noanswer</literal> option will cause the call
+			to be answered and a final 200 response to be sent.
+			</para></warning>
+			<para>
+			</para>
+			<para>As with the <literal>Hangup</literal> application, 
+			the dialplan will terminate after calling this function.</para>
+			<para>
+			</para>
+			<para>The cause code set on the channel will be translated to
+			a standard ISDN cause code using the table defined in
+			ast_sip_hangup_sip2cause() in res_pjsip.c</para>
+			<para>
+			</para>
+			<example title="Terminate call with 437 response code">
+			same = n,PJSIPHangup(437)
+			</example>
+			<example title="Terminate call with 437 response code using the response code name">
+			same = n,PJSIPHangup(UNSUPPORTED_CERTIFICATE)
+			</example>
+			<example title="Terminate call with 437 response code based on condition">
+			same = n,ExecIf($[${SOMEVALUE} = ${SOME_BAD_VALUE}]?PJSIPHangup(437))
+			</example>
+		</description>
+	</application>
+
+	<manager name="PJSIPHangup" language="en_US">
+		<synopsis>
+			Hangup an incoming PJSIP channel with a SIP response code
+		</synopsis>
+		<syntax>
+			<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
+			<xi:include xpointer="xpointer(/docs/manager[@name='Hangup']/syntax/parameter[@name='Channel'])" />
+			<xi:include xpointer="xpointer(/docs/application[@name='PJSIPHangup']/syntax/parameter[@name='Cause'])" />
+		</syntax>
+		<description>
+			<para>
+			Hangs up an incoming PJSIP channel and returns the
+			specified SIP response code in the final response to the caller.
+			</para>
+			<para>
+			</para>
+			<warning><para>
+			This function must be called BEFORE anything that
+			might cause any other final (non 1XX) response to be sent.
+			For example calling <literal>Answer()</literal> or
+			<literal>Playback</literal> without the
+			<literal>noanswer</literal> option will cause the call
+			to be answered and a final 200 response to be sent.
+			</para></warning>
+			<para>
+			</para>
+			<para>The cause code set on the channel will be translated to
+			a standard ISDN cause code using the table defined in
+			ast_sip_hangup_sip2cause() in res_pjsip.c</para>
+			<para>
+			</para>
+			<example title="Terminate call with 437 response code">
+			Action: PJSIPHangup
+			ActionID: 12345678
+			Channel: PJSIP/alice-00000002
+			Cause: 437
+			</example>
+			<example title="Terminate call with 437 response code using the response code name">
+			Action: PJSIPHangup
+			ActionID: 12345678
+			Channel: PJSIP/alice-00000002
+			Cause: UNSUPPORTED_CERTIFICATE
+			</example>
+		</description>
+	</manager>
+
+	<function name="PJSIP_DIAL_CONTACTS" language="en_US">
+		<synopsis>
+			Return a dial string for dialing all contacts on an AOR.
+		</synopsis>
+		<syntax>
+			<parameter name="endpoint" required="true">
+				<para>Name of the endpoint</para>
+			</parameter>
+			<parameter name="aor" required="false">
+				<para>Name of an AOR to use, if not specified the configured AORs on the endpoint are used</para>
+			</parameter>
+			<parameter name="request_user" required="false">
+				<para>Optional request user to use in the request URI</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Returns a properly formatted dial string for dialing all contacts on an AOR.</para>
+		</description>
+	</function>
+	<function name="PJSIP_MEDIA_OFFER" language="en_US">
+		<synopsis>
+			Media and codec offerings to be set on an outbound SIP channel prior to dialing.
+		</synopsis>
+		<syntax>
+			<parameter name="media" required="true">
+				<para>types of media offered</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>When read, returns the codecs offered based upon the media choice.</para>
+			<para>When written, sets the codecs to offer when an outbound dial attempt is made,
+			or when a session refresh is sent using <replaceable>PJSIP_SEND_SESSION_REFRESH</replaceable>.
+			</para>
+		</description>
+		<see-also>
+			<ref type="function">PJSIP_SEND_SESSION_REFRESH</ref>
+		</see-also>
+	</function>
+	<function name="PJSIP_DTMF_MODE" language="en_US">
+		<since>
+			<version>13.18.0</version>
+			<version>14.7.0</version>
+			<version>15.1.0</version>
+			<version>16.0.0</version>
+		</since>
+		<synopsis>
+			Get or change the DTMF mode for a SIP call.
+		</synopsis>
+		<syntax>
+		</syntax>
+		<description>
+			<para>When read, returns the current DTMF mode</para>
+			<para>When written, sets the current DTMF mode</para>
+			<para>This function uses the same DTMF mode naming as the dtmf_mode configuration option</para>
+		</description>
+	</function>
+	<function name="PJSIP_MOH_PASSTHROUGH" language="en_US">
+		<synopsis>
+			Get or change the on-hold behavior for a SIP call.
+		</synopsis>
+		<syntax>
+		</syntax>
+		<description>
+			<para>When read, returns the current moh passthrough mode</para>
+			<para>When written, sets the current moh passthrough mode</para>
+			<para>If <replaceable>yes</replaceable>, on-hold re-INVITEs are sent. If <replaceable>no</replaceable>, music on hold is generated.</para>
+			<para>This function can be used to override the moh_passthrough configuration option</para>
+		</description>
+	</function>
+	<function name="PJSIP_SEND_SESSION_REFRESH" language="en_US">
+		<since>
+			<version>13.12.0</version>
+			<version>14.1.0</version>
+			<version>15.0.0</version>
+		</since>
+		<synopsis>
+			W/O: Initiate a session refresh via an UPDATE or re-INVITE on an established media session
+		</synopsis>
+		<syntax>
+			<parameter name="update_type" required="false">
+				<para>The type of update to send. Default is <literal>invite</literal>.</para>
+				<enumlist>
+					<enum name="invite">
+						<para>Send the session refresh as a re-INVITE.</para>
+					</enum>
+					<enum name="update">
+						<para>Send the session refresh as an UPDATE.</para>
+					</enum>
+				</enumlist>
+			</parameter>
+		</syntax>
+		<description>
+			<para>This function will cause the PJSIP stack to immediately refresh
+			the media session for the channel. This will be done using either a
+			re-INVITE (default) or an UPDATE request.
+			</para>
+			<para>This is most useful when combined with the <replaceable>PJSIP_MEDIA_OFFER</replaceable>
+			dialplan function, as it allows the formats in use on a channel to be
+			re-negotiated after call setup.</para>
+			<warning>
+				<para>The formats the endpoint supports are <emphasis>not</emphasis>
+				checked or enforced by this function. Using this function to offer
+				formats not supported by the endpoint <emphasis>may</emphasis> result
+				in a loss of media.</para>
+			</warning>
+			<example title="Re-negotiate format to g722">
+			 ; Within some existing extension on an answered channel
+			 same => n,Set(PJSIP_MEDIA_OFFER(audio)=!all,g722)
+			 same => n,Set(PJSIP_SEND_SESSION_REFRESH()=invite)
+			</example>
+		</description>
+		<see-also>
+			<ref type="function">PJSIP_MEDIA_OFFER</ref>
+		</see-also>
+	</function>
+	<function name="PJSIP_PARSE_URI" language="en_US">
+		<since>
+			<version>13.24.0</version>
+			<version>16.1.0</version>
+			<version>17.0.0</version>
+		</since>
+		<synopsis>
+			Parse an uri and return a type part of the URI.
+		</synopsis>
+		<syntax>
+			<parameter name="uri" required="true">
+				<para>URI to parse</para>
+			</parameter>
+			<parameter name="type" required="true">
+				<para>The <literal>type</literal> parameter specifies which URI part to read</para>
+				<enumlist>
+					<enum name="display">
+						<para>Display name.</para>
+					</enum>
+					<enum name="scheme">
+						<para>URI scheme.</para>
+					</enum>
+					<enum name="user">
+						<para>User part.</para>
+					</enum>
+					<enum name="passwd">
+						<para>Password part.</para>
+					</enum>
+					<enum name="host">
+						<para>Host part.</para>
+					</enum>
+					<enum name="port">
+						<para>Port number, or zero.</para>
+					</enum>
+					<enum name="user_param">
+						<para>User parameter.</para>
+					</enum>
+					<enum name="method_param">
+						<para>Method parameter.</para>
+					</enum>
+					<enum name="transport_param">
+						<para>Transport parameter.</para>
+					</enum>
+					<enum name="ttl_param">
+						<para>TTL param, or -1.</para>
+					</enum>
+					<enum name="lr_param">
+						<para>Loose routing param, or zero.</para>
+					</enum>
+					<enum name="maddr_param">
+						<para>Maddr param.</para>
+					</enum>
+				</enumlist>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Parse an URI and return a specified part of the URI.</para>
+		</description>
+	</function>
+
+	<info name="CHANNEL" language="en_US" tech="PJSIP">
+		<enumlist>
+			<enum name="rtp">
+				<para>R/O Retrieve media related information.</para>
+				<parameter name="type" required="true">
+					<para>When <replaceable>rtp</replaceable> is specified, the
+					<literal>type</literal> parameter must be provided. It specifies
+					which RTP parameter to read.</para>
+					<enumlist>
+						<enum name="src">
+							<para>Retrieve the local address for RTP.</para>
+						</enum>
+						<enum name="dest">
+							<para>Retrieve the remote address for RTP.</para>
+						</enum>
+						<enum name="direct">
+							<para>If direct media is enabled, this address is the remote address
+							used for RTP.</para>
+						</enum>
+						<enum name="secure">
+							<para>Whether or not the media stream is encrypted.</para>
+							<enumlist>
+								<enum name="0">
+									<para>The media stream is not encrypted.</para>
+								</enum>
+								<enum name="1">
+									<para>The media stream is encrypted.</para>
+								</enum>
+							</enumlist>
+						</enum>
+						<enum name="hold">
+							<para>Whether or not the media stream is currently restricted
+							due to a call hold.</para>
+							<enumlist>
+								<enum name="0">
+									<para>The media stream is not held.</para>
+								</enum>
+								<enum name="1">
+									<para>The media stream is held.</para>
+								</enum>
+							</enumlist>
+						</enum>
+					</enumlist>
+				</parameter>
+				<parameter name="media_type" required="false">
+					<para>When <replaceable>rtp</replaceable> is specified, the
+					<literal>media_type</literal> parameter may be provided. It specifies
+					which media stream the chosen RTP parameter should be retrieved
+					from.</para>
+					<enumlist>
+						<enum name="audio">
+							<para>Retrieve information from the audio media stream.</para>
+							<note><para>If not specified, <literal>audio</literal> is used
+							by default.</para></note>
+						</enum>
+						<enum name="video">
+							<para>Retrieve information from the video media stream.</para>
+						</enum>
+					</enumlist>
+				</parameter>
+			</enum>
+			<enum name="rtcp">
+				<para>R/O Retrieve RTCP statistics.</para>
+				<parameter name="statistic" required="true">
+					<para>When <replaceable>rtcp</replaceable> is specified, the
+					<literal>statistic</literal> parameter must be provided. It specifies
+					which RTCP statistic parameter to read.</para>
+					<enumlist>
+						<enum name="all">
+							<para>Retrieve a summary of all RTCP statistics.</para>
+							<para>The following data items are returned in a semi-colon
+							delineated list:</para>
+							<enumlist>
+								<enum name="ssrc">
+									<para>Our Synchronization Source identifier</para>
+								</enum>
+								<enum name="themssrc">
+									<para>Their Synchronization Source identifier</para>
+								</enum>
+								<enum name="lp">
+									<para>Our lost packet count</para>
+								</enum>
+								<enum name="rxjitter">
+									<para>Received packet jitter</para>
+								</enum>
+								<enum name="rxcount">
+									<para>Received packet count</para>
+								</enum>
+								<enum name="txjitter">
+									<para>Transmitted packet jitter</para>
+								</enum>
+								<enum name="txcount">
+									<para>Transmitted packet count</para>
+								</enum>
+								<enum name="rlp">
+									<para>Remote lost packet count</para>
+								</enum>
+								<enum name="rtt">
+									<para>Round trip time</para>
+								</enum>
+								<enum name="txmes">
+									<para>Transmitted Media Experience Score</para>
+								</enum>
+								<enum name="rxmes">
+									<para>Received Media Experience Score</para>
+								</enum>
+							</enumlist>
+						</enum>
+						<enum name="all_jitter">
+							<para>Retrieve a summary of all RTCP Jitter statistics.</para>
+							<para>The following data items are returned in a semi-colon
+							delineated list:</para>
+							<enumlist>
+								<enum name="minrxjitter">
+									<para>Our minimum jitter</para>
+								</enum>
+								<enum name="maxrxjitter">
+									<para>Our max jitter</para>
+								</enum>
+								<enum name="avgrxjitter">
+									<para>Our average jitter</para>
+								</enum>
+								<enum name="stdevrxjitter">
+									<para>Our jitter standard deviation</para>
+								</enum>
+								<enum name="reported_minjitter">
+									<para>Their minimum jitter</para>
+								</enum>
+								<enum name="reported_maxjitter">
+									<para>Their max jitter</para>
+								</enum>
+								<enum name="reported_avgjitter">
+									<para>Their average jitter</para>
+								</enum>
+								<enum name="reported_stdevjitter">
+									<para>Their jitter standard deviation</para>
+								</enum>
+							</enumlist>
+						</enum>
+						<enum name="all_loss">
+							<para>Retrieve a summary of all RTCP packet loss statistics.</para>
+							<para>The following data items are returned in a semi-colon
+							delineated list:</para>
+							<enumlist>
+								<enum name="minrxlost">
+									<para>Our minimum lost packets</para>
+								</enum>
+								<enum name="maxrxlost">
+									<para>Our max lost packets</para>
+								</enum>
+								<enum name="avgrxlost">
+									<para>Our average lost packets</para>
+								</enum>
+								<enum name="stdevrxlost">
+									<para>Our lost packets standard deviation</para>
+								</enum>
+								<enum name="reported_minlost">
+									<para>Their minimum lost packets</para>
+								</enum>
+								<enum name="reported_maxlost">
+									<para>Their max lost packets</para>
+								</enum>
+								<enum name="reported_avglost">
+									<para>Their average lost packets</para>
+								</enum>
+								<enum name="reported_stdevlost">
+									<para>Their lost packets standard deviation</para>
+								</enum>
+							</enumlist>
+						</enum>
+						<enum name="all_rtt">
+							<para>Retrieve a summary of all RTCP round trip time information.</para>
+							<para>The following data items are returned in a semi-colon
+							delineated list:</para>
+							<enumlist>
+								<enum name="minrtt">
+									<para>Minimum round trip time</para>
+								</enum>
+								<enum name="maxrtt">
+									<para>Maximum round trip time</para>
+								</enum>
+								<enum name="avgrtt">
+									<para>Average round trip time</para>
+								</enum>
+								<enum name="stdevrtt">
+									<para>Standard deviation round trip time</para>
+								</enum>
+							</enumlist>
+						</enum>
+						<enum name="all_mes">
+							<para>Retrieve a summary of all RTCP Media Experience Score information.</para>
+							<para>The following data items are returned in a semi-colon
+							delineated list:</para>
+							<enumlist>
+								<enum name="minmes">
+									<para>Minimum MES based on us analysing received packets.</para>
+								</enum>
+								<enum name="maxmes">
+									<para>Maximum MES based on us analysing received packets.</para>
+								</enum>
+								<enum name="avgmes">
+									<para>Average MES based on us analysing received packets.</para>
+								</enum>
+								<enum name="stdevmes">
+									<para>Standard deviation MES based on us analysing received packets.</para>
+								</enum>
+								<enum name="reported_minmes">
+									<para>Minimum MES based on data we get in Sender and Receiver Reports sent by the remote end</para>
+								</enum>
+								<enum name="reported_maxmes">
+									<para>Maximum MES based on data we get in Sender and Receiver Reports sent by the remote end</para>
+								</enum>
+								<enum name="reported_avgmes">
+									<para>Average MES based on data we get in Sender and Receiver Reports sent by the remote end</para>
+								</enum>
+								<enum name="reported_stdevmes">
+									<para>Standard deviation MES based on data we get in Sender and Receiver Reports sent by the remote end</para>
+								</enum>
+							</enumlist>
+						</enum>
+						<enum name="txcount"><para>Transmitted packet count</para></enum>
+						<enum name="rxcount"><para>Received packet count</para></enum>
+						<enum name="txjitter"><para>Transmitted packet jitter</para></enum>
+						<enum name="rxjitter"><para>Received packet jitter</para></enum>
+						<enum name="remote_maxjitter"><para>Their max jitter</para></enum>
+						<enum name="remote_minjitter"><para>Their minimum jitter</para></enum>
+						<enum name="remote_normdevjitter"><para>Their average jitter</para></enum>
+						<enum name="remote_stdevjitter"><para>Their jitter standard deviation</para></enum>
+						<enum name="local_maxjitter"><para>Our max jitter</para></enum>
+						<enum name="local_minjitter"><para>Our minimum jitter</para></enum>
+						<enum name="local_normdevjitter"><para>Our average jitter</para></enum>
+						<enum name="local_stdevjitter"><para>Our jitter standard deviation</para></enum>
+						<enum name="txploss"><para>Transmitted packet loss</para></enum>
+						<enum name="rxploss"><para>Received packet loss</para></enum>
+						<enum name="remote_maxrxploss"><para>Their max lost packets</para></enum>
+						<enum name="remote_minrxploss"><para>Their minimum lost packets</para></enum>
+						<enum name="remote_normdevrxploss"><para>Their average lost packets</para></enum>
+						<enum name="remote_stdevrxploss"><para>Their lost packets standard deviation</para></enum>
+						<enum name="local_maxrxploss"><para>Our max lost packets</para></enum>
+						<enum name="local_minrxploss"><para>Our minimum lost packets</para></enum>
+						<enum name="local_normdevrxploss"><para>Our average lost packets</para></enum>
+						<enum name="local_stdevrxploss"><para>Our lost packets standard deviation</para></enum>
+						<enum name="rtt"><para>Round trip time</para></enum>
+						<enum name="maxrtt"><para>Maximum round trip time</para></enum>
+						<enum name="minrtt"><para>Minimum round trip time</para></enum>
+						<enum name="normdevrtt"><para>Average round trip time</para></enum>
+						<enum name="stdevrtt"><para>Standard deviation round trip time</para></enum>
+						<enum name="local_ssrc"><para>Our Synchronization Source identifier</para></enum>
+						<enum name="remote_ssrc"><para>Their Synchronization Source identifier</para></enum>
+						<enum name="txmes"><para>
+						Current MES based on us analyzing rtt, jitter and loss
+						in the actual received RTP stream received from the remote end.
+						I.E.  This is the MES for the incoming audio stream.
+						</para></enum>
+						<enum name="rxmes"><para>
+						Current MES based on rtt and the jitter and loss values in
+						RTCP sender and receiver reports we receive from the
+						remote end. I.E.  This is the MES for the outgoing audio stream.
+						</para></enum>
+						<enum name="remote_maxmes"><para>Max MES based on data we get in Sender and Receiver Reports sent by the remote end</para></enum>
+						<enum name="remote_minmes"><para>Min MES based on data we get in Sender and Receiver Reports sent by the remote end</para></enum>
+						<enum name="remote_normdevmes"><para>Average MES based on data we get in Sender and Receiver Reports sent by the remote end</para></enum>
+						<enum name="remote_stdevmes"><para>Standard deviation MES based on data we get in Sender and Receiver Reports sent by the remote end</para></enum>
+						<enum name="local_maxmes"><para>Max MES based on us analyzing the received RTP stream</para></enum>
+						<enum name="local_minmes"><para>Min MES based on us analyzing the received RTP stream</para></enum>
+						<enum name="local_normdevmes"><para>Average MES based on us analyzing the received RTP stream</para></enum>
+						<enum name="local_stdevmes"><para>Standard deviation MES based on us analyzing the received RTP stream</para></enum>
+					</enumlist>
+				</parameter>
+				<parameter name="media_type" required="false">
+					<para>When <replaceable>rtcp</replaceable> is specified, the
+					<literal>media_type</literal> parameter may be provided. It specifies
+					which media stream the chosen RTCP parameter should be retrieved
+					from.</para>
+					<enumlist>
+						<enum name="audio">
+							<para>Retrieve information from the audio media stream.</para>
+							<note><para>If not specified, <literal>audio</literal> is used
+							by default.</para></note>
+						</enum>
+						<enum name="video">
+							<para>Retrieve information from the video media stream.</para>
+						</enum>
+					</enumlist>
+				</parameter>
+			</enum>
+			<enum name="endpoint">
+				<para>R/O The name of the endpoint associated with this channel.
+				Use the <replaceable>PJSIP_ENDPOINT</replaceable> function to obtain
+				further endpoint related information.</para>
+			</enum>
+			<enum name="contact">
+				<para>R/O The name of the contact associated with this channel.
+				Use the <replaceable>PJSIP_CONTACT</replaceable> function to obtain
+				further contact related information. Note this may not be present and if so
+				is only available on outgoing legs.</para>
+			</enum>
+			<enum name="aor">
+				<para>R/O The name of the AOR associated with this channel.
+				Use the <replaceable>PJSIP_AOR</replaceable> function to obtain
+				further AOR related information. Note this may not be present and if so
+				is only available on outgoing legs.</para>
+			</enum>
+			<enum name="pjsip">
+				<para>R/O Obtain information about the current PJSIP channel and its
+				session.</para>
+				<parameter name="type" required="true">
+					<para>When <replaceable>pjsip</replaceable> is specified, the
+					<literal>type</literal> parameter must be provided. It specifies
+					which signalling parameter to read.</para>
+					<enumlist>
+						<enum name="call-id">
+							<para>The SIP call-id.</para>
+						</enum>
+						<enum name="secure">
+							<para>Whether or not the signalling uses a secure transport.</para>
+							<enumlist>
+								<enum name="0"><para>The signalling uses a non-secure transport.</para></enum>
+								<enum name="1"><para>The signalling uses a secure transport.</para></enum>
+							</enumlist>
+						</enum>
+						<enum name="target_uri">
+							<para>The contact URI where requests are sent.</para>
+						</enum>
+						<enum name="local_uri">
+							<para>The local URI.</para>
+						</enum>
+						<enum name="local_tag">
+							<para>Tag in From header</para>
+						</enum>
+						<enum name="remote_uri">
+							<para>The remote URI.</para>
+						</enum>
+						<enum name="remote_tag">
+							<para>Tag in To header</para>
+						</enum>
+						<enum name="request_uri">
+							<para>The request URI of the incoming <literal>INVITE</literal>
+							associated with the creation of this channel.</para>
+						</enum>
+						<enum name="t38state">
+							<para>The current state of any T.38 fax on this channel.</para>
+							<enumlist>
+								<enum name="DISABLED"><para>T.38 faxing is disabled on this channel.</para></enum>
+								<enum name="LOCAL_REINVITE"><para>Asterisk has sent a <literal>re-INVITE</literal> to the remote end to initiate a T.38 fax.</para></enum>
+								<enum name="REMOTE_REINVITE"><para>The remote end has sent a <literal>re-INVITE</literal> to Asterisk to initiate a T.38 fax.</para></enum>
+								<enum name="ENABLED"><para>A T.38 fax session has been enabled.</para></enum>
+								<enum name="REJECTED"><para>A T.38 fax session was attempted but was rejected.</para></enum>
+							</enumlist>
+						</enum>
+						<enum name="local_addr">
+							<para>On inbound calls, the full IP address and port number that
+							the <literal>INVITE</literal> request was received on. On outbound
+							calls, the full IP address and port number that the <literal>INVITE</literal>
+							request was transmitted from.</para>
+						</enum>
+						<enum name="remote_addr">
+							<para>On inbound calls, the full IP address and port number that
+							the <literal>INVITE</literal> request was received from. On outbound
+							calls, the full IP address and port number that the <literal>INVITE</literal>
+							request was transmitted to.</para>
+						</enum>
+					</enumlist>
+				</parameter>
+			</enum>
+		</enumlist>
+	</info>
+	<info name="CHANNEL_EXAMPLES" language="en_US" tech="PJSIP">
+		<example title="PJSIP specific CHANNEL examples">
+			; Log the current Call-ID
+			same => n,Log(NOTICE, ${CHANNEL(pjsip,call-id)})
+	
+			; Log the destination address of the audio stream
+			same => n,Log(NOTICE, ${CHANNEL(rtp,dest)})
+	
+			; Store the round-trip time associated with a
+			; video stream in the CDR field video-rtt
+			same => n,Set(CDR(video-rtt)=${CHANNEL(rtcp,rtt,video)})
+		</example>
+	</info>
+</docs>

channels/pjsip/include/dialplan_functions.h

@@ -148,4 +148,24 @@ int pjsip_acf_dial_contacts_read(struct ast_channel *chan, const char *cmd, char
  */
 int pjsip_acf_parse_uri_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len);
 
-#endif /* _PJSIP_DIALPLAN_FUNCTIONS */
+/*!
+ * \brief Hang up an incoming PJSIP channel with a SIP response code
+ * \param chan The channel the function is called on
+ * \param data SIP response code or name
+ *
+ * \retval 0 on success
+ * \retval -1 on failure
+ */
+int pjsip_app_hangup(struct ast_channel *chan, const char *data);
+
+/*!
+ * \brief Manager action to hang up an incoming PJSIP channel with a SIP response code
+ * \param s session
+ * \param m message
+ *
+ * \retval 0 on success
+ * \retval -1 on failure
+ */
+int pjsip_action_hangup(struct mansession *s, const struct message *m);
+
+#endif /* _PJSIP_DIALPLAN_FUNCTIONS */

channels/sig_analog.c

@@ -805,6 +805,11 @@ int analog_available(struct analog_pvt *p)
 		return 0;
 	}
 
+	/* If line is being held, definitely not (don't allow call waitings to an on-hook phone) */
+	if (p->cshactive) {
+		return 0;
+	}
+
 	/* If no owner definitely available */
 	if (!p->owner) {
 		offhook = analog_is_off_hook(p);
@@ -1300,6 +1305,7 @@ int analog_hangup(struct analog_pvt *p, struct ast_channel *ast)
 		p->channel, idx, p->subs[ANALOG_SUB_REAL].allocd, p->subs[ANALOG_SUB_CALLWAIT].allocd, p->subs[ANALOG_SUB_THREEWAY].allocd);
 	if (idx > -1) {
 		/* Real channel, do some fixup */
+		p->cshactive = 0;
 		p->subs[idx].owner = NULL;
 		p->polarity = POLARITY_IDLE;
 		analog_set_linear_mode(p, idx, 0);
@@ -1758,10 +1764,7 @@ static void *__analog_ss_thread(void *data)
 
 	ast_debug(1, "%s %d\n", __FUNCTION__, p->channel);
 
-	if (!chan) {
-		/* What happened to the channel? */
-		goto quit;
-	}
+	ast_assert(chan != NULL);
 
 	if ((callid = ast_channel_callid(chan))) {
 		ast_callid_threadassoc_add(callid);
@@ -2750,6 +2753,7 @@ int analog_ss_thread_start(struct analog_pvt *p, struct ast_channel *chan)
 {
 	pthread_t threadid;
 
+	p->ss_astchan = chan;
 	return ast_pthread_create_detached(&threadid, NULL, __analog_ss_thread, p);
 }
 
@@ -2933,6 +2937,34 @@ static struct ast_frame *__analog_handle_event(struct analog_pvt *p, struct ast_
 		analog_get_and_handle_alarms(p);
 		cause_code->ast_cause = AST_CAUSE_NETWORK_OUT_OF_ORDER;
 	case ANALOG_EVENT_ONHOOK:
+		if (p->calledsubscriberheld && (p->sig == ANALOG_SIG_FXOLS || p->sig == ANALOG_SIG_FXOGS || p->sig == ANALOG_SIG_FXOKS) && idx == ANALOG_SUB_REAL) {
+			ast_debug(4, "Channel state on %s is %d\n", ast_channel_name(ast), ast_channel_state(ast));
+			/* Called Subscriber Held: don't let the called party hang up on an incoming call immediately (if it's the only call). */
+			if (p->subs[ANALOG_SUB_CALLWAIT].owner || p->subs[ANALOG_SUB_THREEWAY].owner) {
+				ast_debug(2, "Letting this call hang up normally, since it's not the only call\n");
+			} else if (!p->owner || !p->subs[ANALOG_SUB_REAL].owner || ast_channel_state(ast) != AST_STATE_UP) {
+				ast_debug(2, "Called Subscriber Held does not apply: channel state is %d\n", ast_channel_state(ast));
+			} else if (!p->owner || !p->subs[ANALOG_SUB_REAL].owner || strcmp(ast_channel_appl(p->subs[ANALOG_SUB_REAL].owner), "AppDial")) {
+				/* Called Subscriber held only applies to incoming calls, not outgoing calls.
+				 * We can't use p->outgoing because that is always true, for both incoming and outgoing calls, so it's not accurate.
+				 * We can check the channel application/data instead.
+				 * For incoming calls to the channel, it will look like: AppDial / (Outgoing Line)
+				 * We only want this behavior for regular calls anyways (and not, say, Queue),
+				 * so this would actually work great. But accessing ast_channel_appl can cause a crash if there are no calls left,
+				 * so this check must occur AFTER we confirm the channel state *is* still UP.
+				 */
+				ast_debug(2, "Called Subscriber Held does not apply: not an incoming call\n");
+			} else if (analog_is_off_hook(p)) {
+				ast_log(LOG_WARNING, "Got ONHOOK but channel %d is off hook?\n", p->channel); /* Shouldn't happen */
+			} else {
+				ast_verb(3, "Holding incoming call %s for channel %d\n", ast_channel_name(ast), p->channel);
+				/* Inhibit dahdi_hangup from getting called, and do nothing else now.
+				 * When the DAHDI channel goes off hook again, it'll just get reconnected with the incoming call,
+				 * to which, as far as its concerned, nothing has happened. */
+				p->cshactive = 1; /* Keep track that this DAHDI channel is currently being held by an incoming call. */
+				break;
+			}
+		}
 		ast_queue_control_data(ast, AST_CONTROL_PVT_CAUSE_CODE, cause_code, data_size);
 		ast_channel_hangupcause_hash_set(ast, cause_code, data_size);
 		switch (p->sig) {
@@ -3809,6 +3841,7 @@ void *analog_handle_init_event(struct analog_pvt *i, int event)
 		case ANALOG_SIG_FXOKS:
 			res = analog_off_hook(i);
 			i->fxsoffhookstate = 1;
+			i->cshactive = 0;
 			if (res && (errno == EBUSY)) {
 				break;
 			}

channels/sig_analog.h

@@ -289,6 +289,7 @@ struct analog_pvt {
 	unsigned int ani_wink_time:16;			/* Safe wait time before we wink to start ANI spill */
 
 	unsigned int answeronpolarityswitch:1;
+	unsigned int calledsubscriberheld:1;	/*!< TRUE if a single incoming call can hold an FXS channel */
 	unsigned int callreturn:1;
 	unsigned int cancallforward:1;
 	unsigned int canpark:1;
@@ -330,6 +331,7 @@ struct analog_pvt {
 
 	/* XXX: All variables after this are internal */
 	unsigned int callwaiting:1;		/*!< TRUE if call waiting is enabled. (Active option) */
+	unsigned int cshactive:1;		/*!< TRUE if FXS channel is currently held by an incoming call */
 	unsigned int dialednone:1;
 	unsigned int dialing:1;			/*!< TRUE if in the process of dialing digits or sending something */
 	unsigned int dnd:1;				/*!< TRUE if Do-Not-Disturb is enabled. */

codecs/codec_speex.c

@@ -597,20 +597,20 @@ static int parse_config(int reload)
 		if (!strcasecmp(var->name, "quality")) {
 			res = abs(atoi(var->value));
 			if (res > -1 && res < 11) {
-				ast_verb(3, "CODEC SPEEX: Setting Quality to %d\n",res);
+				ast_verb(5, "CODEC SPEEX: Setting Quality to %d\n",res);
 				quality = res;
 			} else
 				ast_log(LOG_ERROR,"Error Quality must be 0-10\n");
 		} else if (!strcasecmp(var->name, "complexity")) {
 			res = abs(atoi(var->value));
 			if (res > -1 && res < 11) {
-				ast_verb(3, "CODEC SPEEX: Setting Complexity to %d\n",res);
+				ast_verb(5, "CODEC SPEEX: Setting Complexity to %d\n",res);
 				complexity = res;
 			} else
 				ast_log(LOG_ERROR,"Error! Complexity must be 0-10\n");
 		} else if (!strcasecmp(var->name, "vbr_quality")) {
 			if (sscanf(var->value, "%30f", &res_f) == 1 && res_f >= 0 && res_f <= 10) {
-				ast_verb(3, "CODEC SPEEX: Setting VBR Quality to %f\n",res_f);
+				ast_verb(5, "CODEC SPEEX: Setting VBR Quality to %f\n",res_f);
 				vbr_quality = res_f;
 			} else
 				ast_log(LOG_ERROR,"Error! VBR Quality must be 0-10\n");
@@ -618,62 +618,62 @@ static int parse_config(int reload)
 			ast_log(LOG_ERROR,"Error! ABR Quality setting obsolete, set ABR to desired bitrate\n");
 		} else if (!strcasecmp(var->name, "enhancement")) {
 			enhancement = ast_true(var->value) ? 1 : 0;
-			ast_verb(3, "CODEC SPEEX: Perceptual Enhancement Mode. [%s]\n",enhancement ? "on" : "off");
+			ast_verb(5, "CODEC SPEEX: Perceptual Enhancement Mode. [%s]\n",enhancement ? "on" : "off");
 		} else if (!strcasecmp(var->name, "vbr")) {
 			vbr = ast_true(var->value) ? 1 : 0;
-			ast_verb(3, "CODEC SPEEX: VBR Mode. [%s]\n",vbr ? "on" : "off");
+			ast_verb(5, "CODEC SPEEX: VBR Mode. [%s]\n",vbr ? "on" : "off");
 		} else if (!strcasecmp(var->name, "abr")) {
 			res = abs(atoi(var->value));
 			if (res >= 0) {
 					if (res > 0)
-					ast_verb(3, "CODEC SPEEX: Setting ABR target bitrate to %d\n",res);
+					ast_verb(5, "CODEC SPEEX: Setting ABR target bitrate to %d\n",res);
 					else
-					ast_verb(3, "CODEC SPEEX: Disabling ABR\n");
+					ast_verb(5, "CODEC SPEEX: Disabling ABR\n");
 				abr = res;
 			} else
 				ast_log(LOG_ERROR,"Error! ABR target bitrate must be >= 0\n");
 		} else if (!strcasecmp(var->name, "vad")) {
 			vad = ast_true(var->value) ? 1 : 0;
-			ast_verb(3, "CODEC SPEEX: VAD Mode. [%s]\n",vad ? "on" : "off");
+			ast_verb(5, "CODEC SPEEX: VAD Mode. [%s]\n",vad ? "on" : "off");
 		} else if (!strcasecmp(var->name, "dtx")) {
 			dtx = ast_true(var->value) ? 1 : 0;
-			ast_verb(3, "CODEC SPEEX: DTX Mode. [%s]\n",dtx ? "on" : "off");
+			ast_verb(5, "CODEC SPEEX: DTX Mode. [%s]\n",dtx ? "on" : "off");
 		} else if (!strcasecmp(var->name, "preprocess")) {
 			preproc = ast_true(var->value) ? 1 : 0;
-			ast_verb(3, "CODEC SPEEX: Preprocessing. [%s]\n",preproc ? "on" : "off");
+			ast_verb(5, "CODEC SPEEX: Preprocessing. [%s]\n",preproc ? "on" : "off");
 		} else if (!strcasecmp(var->name, "pp_vad")) {
 			pp_vad = ast_true(var->value) ? 1 : 0;
-			ast_verb(3, "CODEC SPEEX: Preprocessor VAD. [%s]\n",pp_vad ? "on" : "off");
+			ast_verb(5, "CODEC SPEEX: Preprocessor VAD. [%s]\n",pp_vad ? "on" : "off");
 		} else if (!strcasecmp(var->name, "pp_agc")) {
 			pp_agc = ast_true(var->value) ? 1 : 0;
-			ast_verb(3, "CODEC SPEEX: Preprocessor AGC. [%s]\n",pp_agc ? "on" : "off");
+			ast_verb(5, "CODEC SPEEX: Preprocessor AGC. [%s]\n",pp_agc ? "on" : "off");
 		} else if (!strcasecmp(var->name, "pp_agc_level")) {
 			if (sscanf(var->value, "%30f", &res_f) == 1 && res_f >= 0) {
-				ast_verb(3, "CODEC SPEEX: Setting preprocessor AGC Level to %f\n",res_f);
+				ast_verb(5, "CODEC SPEEX: Setting preprocessor AGC Level to %f\n",res_f);
 				pp_agc_level = res_f;
 			} else
 				ast_log(LOG_ERROR,"Error! Preprocessor AGC Level must be >= 0\n");
 		} else if (!strcasecmp(var->name, "pp_denoise")) {
 			pp_denoise = ast_true(var->value) ? 1 : 0;
-			ast_verb(3, "CODEC SPEEX: Preprocessor Denoise. [%s]\n",pp_denoise ? "on" : "off");
+			ast_verb(5, "CODEC SPEEX: Preprocessor Denoise. [%s]\n",pp_denoise ? "on" : "off");
 		} else if (!strcasecmp(var->name, "pp_dereverb")) {
 			pp_dereverb = ast_true(var->value) ? 1 : 0;
-			ast_verb(3, "CODEC SPEEX: Preprocessor Dereverb. [%s]\n",pp_dereverb ? "on" : "off");
+			ast_verb(5, "CODEC SPEEX: Preprocessor Dereverb. [%s]\n",pp_dereverb ? "on" : "off");
 		} else if (!strcasecmp(var->name, "pp_dereverb_decay")) {
 			if (sscanf(var->value, "%30f", &res_f) == 1 && res_f >= 0) {
-				ast_verb(3, "CODEC SPEEX: Setting preprocessor Dereverb Decay to %f\n",res_f);
+				ast_verb(5, "CODEC SPEEX: Setting preprocessor Dereverb Decay to %f\n",res_f);
 				pp_dereverb_decay = res_f;
 			} else
 				ast_log(LOG_ERROR,"Error! Preprocessor Dereverb Decay must be >= 0\n");
 		} else if (!strcasecmp(var->name, "pp_dereverb_level")) {
 			if (sscanf(var->value, "%30f", &res_f) == 1 && res_f >= 0) {
-				ast_verb(3, "CODEC SPEEX: Setting preprocessor Dereverb Level to %f\n",res_f);
+				ast_verb(5, "CODEC SPEEX: Setting preprocessor Dereverb Level to %f\n",res_f);
 				pp_dereverb_level = res_f;
 			} else
 				ast_log(LOG_ERROR,"Error! Preprocessor Dereverb Level must be >= 0\n");
 		} else if (!strcasecmp(var->name, "experimental_rtcp_feedback")) {
 			exp_rtcp_fb = ast_true(var->value) ? 1 : 0;
-			ast_verb(3, "CODEC SPEEX: Experimental RTCP Feedback. [%s]\n",exp_rtcp_fb ? "on" : "off");
+			ast_verb(5, "CODEC SPEEX: Experimental RTCP Feedback. [%s]\n",exp_rtcp_fb ? "on" : "off");
 		}
 	}
 	ast_config_destroy(cfg);

configs/basic-pbx/README

@@ -8,8 +8,8 @@ If you intend to use this configuration as a template for your own, then
 you will need to change many values in the various configuration files to
 match your own devices, network, SIP ITSP accounts and more.
 
-For further documentation on this configuration see the Asterisk wiki:
-https://wiki.asterisk.org/wiki/display/AST/Reference+Use+Cases+for+Asterisk.
+For further documentation on this configuration see the Asterisk documentation:
+https://docs.asterisk.org/Deployment/Reference-Use-Cases-for-Asterisk/.
 
 Please report bugs or errors in configuration on the Asterisk issue tracker:
-https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
+https://docs.asterisk.org/Asterisk-Community/Asterisk-Issue-Guidelines/

configs/samples/asterisk.conf.sample

@@ -126,6 +126,10 @@ documentation_language = en_US	; Set the language you want documentation
                 ; housekeeping AMI and ARI channel events.  This can
                 ; reduce the load on the manager and ARI applications
                 ; when the Digium Phone Module for Asterisk is in use.
+;sounds_search_custom_dir = no;  This option, if enabled, will
+                ; cause Asterisk to search for sounds files in
+                ; AST_DATA_DIR/sounds/custom before searching the
+                ; normal directories like AST_DATA_DIR/sounds/<lang>.
 
 ; Changing the following lines may compromise your security.
 ;[files]

configs/samples/ccss.conf.sample

@@ -2,7 +2,7 @@
 ; --- Call Completion Supplementary Services ---
 ;
 ; For more information about CCSS, see the CCSS user documentation
-; https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+(CCSS)
+; https://docs.asterisk.org/Deployment/PSTN-Connectivity/Call-Completion-Supplementary-Services-CCSS/
 ;
 
 [general]

configs/samples/chan_dahdi.conf.sample

@@ -595,7 +595,7 @@ usecallerid=yes
 ;     polarity    = polarity reversal signals the start
 ;     polarity_IN = polarity reversal signals the start, for India,
 ;                   for dtmf dialtone detection; using DTMF.
-;     (see https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India)
+; (see https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India)
 ;     dtmf        = causes monitor loop to look for dtmf energy on the
 ;                   incoming channel to initate cid acquisition
 ;
@@ -755,6 +755,18 @@ usecallingpres=yes
 ;
 callwaitingcallerid=yes
 ;
+; Whether or not to allow users to go on-hook when receiving an incoming call
+; without disconnecting it. Users can later resume the call from any phone
+; on the same physical phone line (the same DAHDI channel).
+; This setting only has an effect on FXS (FXO-signalled) channels where there
+; is only a single incoming call to the DAHDI channel, using the Dial application.
+; (This is a convenience mechanism to avoid users wishing to resume a conversation
+; at a different phone from leaving a phone off the hook, resuming elsewhere,
+; and forgetting to restore the original phone on hook afterwards.)
+; Default is no.
+;
+;calledsubscriberheld=yes
+;
 ; Support three-way calling
 ;
 threewaycalling=yes
@@ -932,6 +944,10 @@ group=1
 ; you can answer it by picking up and dialing *8#.  For simple offices, just
 ; make these both the same.  Groups range from 0 to 63.
 ;
+; Call groups and pickup groups may only be specified for FXO signalled channels.
+; If you need to pick up an FXS signalled channel directly, you can have it
+; dial a Local channel and pick up the ;1 side of the Local channel instead.
+;
 callgroup=1
 pickupgroup=1
 ;
@@ -1563,7 +1579,7 @@ pickupgroup=1
 ;#include ss7.timers
 
 ; For more information on setting up SS7, see the README file in libss7 or
-; https://wiki.asterisk.org/wiki/display/AST/Signaling+System+Number+7
+; https://docs.asterisk.org/Deployment/PSTN-Connectivity/Signaling-System-Number-7/
 ; ----------------- SS7 Options ----------------------------------------
 
 ; ---------------- Options for use with signalling=mfcr2 --------------

configs/samples/dundi.conf.sample

@@ -29,7 +29,7 @@
 ;bindaddr=0.0.0.0
 ;port=4520
 ;
-; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of the tos parameter.
+; See https://docs.asterisk.org/Configuration/Channel-Drivers/IP-Quality-of-Service for a description of the tos parameter.
 ;tos=ef
 ;
 ; Our entity identifier.  (It should generally be the MAC address of the

configs/samples/extconfig.conf.sample

@@ -2,7 +2,7 @@
 ; Static and realtime external configuration
 ; engine configuration
 ;
-; See https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
+; See https://docs.asterisk.org/Fundamentals/Asterisk-Configuration/Database-Support-Configuration/Realtime-Database-Configuration/
 ; for basic table formatting information.
 ;
 [settings]
@@ -84,6 +84,7 @@
 ;ps_outbound_publishes => odbc,asterisk
 ;ps_inbound_publications = odbc,asterisk
 ;ps_asterisk_publications = odbc,asterisk
+;stir_tn => odbc,asterisk
 ;voicemail => odbc,asterisk
 ;extensions => odbc,asterisk
 ;meetme => mysql,general

configs/samples/geolocation.conf.sample

@@ -1,7 +1,7 @@
 ;--
   Geolocation Profile Sample Configuration
 
-  Please see https://wiki.asterisk.org/wiki/display/AST/Geolocation
+  Please see https://docs.asterisk.org/Deployment/Geolocation/
   for the most current information.
 --;
 
@@ -33,7 +33,7 @@ incoming calls (Asterisk is the UAS) and and one for outgoing calls
 
 NOTE:
 
-See https://wiki.asterisk.org/wiki/display/AST/Geolocation for the most
+See https://docs.asterisk.org/Deployment/Geolocation/ for the most
 complete and up-to-date information on valid values for the object
 parameters and a full list of references.
 
@@ -96,7 +96,7 @@ variables like ${EXTEN}, channel variables you may have added in the
 dialplan, or variables you may have specified in the profile that
 references this location object.
 
-NOTE: See https://wiki.asterisk.org/wiki/display/AST/Geolocation for the
+NOTE: See https://docs.asterisk.org/Deployment/Geolocation/ for the
 most complete and up-to-date information on valid values for the object
 parameters and a full list of references.
 

configs/samples/iax.conf.sample

@@ -140,9 +140,13 @@
 
 ;
 ; Specify bandwidth of low, medium, or high to control which codecs are used
-; in general.
+; in general. This setting will restrict codecs used to only those that comply
+; with the bandwidth setting. In most cases, you should set this to 'high' so
+; that high-quality codecs may be used; if set to a lower value, this will
+; degrade call quality, so you probably only want to do this if you have
+; actual significant bandwidth constraints.
 ;
-bandwidth=low
+bandwidth=high
 ;
 
 ;
@@ -323,7 +327,7 @@ encryption=yes
 ;
 ;authdebug = yes
 ;
-; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
+; See https://docs.asterisk.org/Configuration/Channel-Drivers/IP-Quality-of-Service for a description of these parameters.
 ;tos=ef
 ;cos=5
 ;

configs/samples/iaxprov.conf.sample

@@ -53,7 +53,7 @@ codec=ulaw
 ;
 flags=register,heartbeat
 ;
-; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of this parameter.
+; See https://docs.asterisk.org/Configuration/Channel-Drivers/IP-Quality-of-Service for a description of this parameter.
 ;tos=ef
 ;
 ; Example iaxy provisioning

configs/samples/pjproject.conf.sample

@@ -38,6 +38,10 @@
 ;  - 5: trace
 ;  - 6: more detailed trace
 ;
+; Note:  setting the pjproject debug level to 4 (debug) or above may result in
+; raw packets being logged. This should only be enabled during active debugging
+; to avoid a potential security issue due to logging injection.
+;
 ;asterisk_error =    ; A comma separated list of pjproject log levels to map to
                      ; Asterisk errors.
                      ; (default: "0,1")

configs/samples/pjsip.conf.sample

@@ -20,7 +20,7 @@
 
 ; Documentation
 ;
-; The official documentation is at http://wiki.asterisk.org
+; The official documentation is at https://docs.asterisk.org
 ; You can read the XML configuration help via Asterisk command line with
 ; "config show help res_pjsip", then you can drill down through the various
 ; sections and their options.
@@ -31,8 +31,8 @@
 ; At a minimum please read the file "README-SERIOUSLY.bestpractices.txt",
 ; located in the Asterisk source directory before starting Asterisk.
 ; Otherwise you risk allowing the security of the Asterisk system to be
-; compromised. Beyond that please visit and read the security information on
-; the wiki at: https://wiki.asterisk.org/wiki/x/EwFB
+; compromised. Beyond that please visit and read the security information in
+; the documentation at: https://docs.asterisk.org/Deployment/Important-Security-Considerations/
 ;
 ; A few basics to pay attention to:
 ;
@@ -47,7 +47,7 @@
 ;
 ; See the example ACL configuration in this file. Read the configuration help
 ; for the section and all of its options. Look over the samples in acl.conf
-; and documentation at https://wiki.asterisk.org/wiki/x/uA80AQ
+; and documentation at https://docs.asterisk.org/Configuration/Core-Configuration/Named-ACLs/
 ; If possible, restrict access to only networks and addresses you trust.
 ;
 ; Dialplan Contexts
@@ -175,7 +175,7 @@
 ;
 ; This is a simple registration that works with some SIP trunking providers.
 ; You'll need to set up the auth example "mytrunk_auth" below to enable outbound
-; authentication. Note that we "outbound_auth=" use for outbound authentication
+; authentication. Note that we use "outbound_auth=" for outbound authentication
 ; instead of "auth=", which is for inbound authentication.
 ;
 ; If you are registering to a server from behind NAT, be sure you assign a transport
@@ -393,7 +393,7 @@
 ;rewrite_contact=yes  ; necessary if endpoint does not know/register public ip:port
 ;ice_support=yes   ;This is specific to clients that support NAT traversal
                    ;for media via ICE,STUN,TURN. See the wiki at:
-                   ;https://wiki.asterisk.org/wiki/x/D4FHAQ
+                   ;https://docs.asterisk.org/Configuration/Miscellaneous/Interactive-Connectivity-Establishment-ICE-in-Asterisk/
                    ;for a deeper explanation of this topic.
 
 ;[6002]
@@ -1454,7 +1454,7 @@
 
 ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_outbound_publish
 ;======================OUTBOUND_PUBLISH SECTION OPTIONS=====================
-; See https://wiki.asterisk.org/wiki/display/AST/Publishing+Extension+State
+; See https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Publishing-Extension-State/
 ; for more information.
 ;[outbound-publish]
 ;type=outbound-publish     ; Must be of type 'outbound-publish'.
@@ -1509,9 +1509,9 @@
 
 
 ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_pubsub
-;=============================RESOURCE-LIST===================================
-; See https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30278158
+; See https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Resource-List-Subscriptions-RLS/
 ; for more information.
+;=============================RESOURCE-LIST===================================
 ;[resource_list]
 ;type=resource_list        ; Must be of type 'resource_list'.
 
@@ -1568,7 +1568,7 @@
 
 
 ;==========================INBOUND_PUBLICATION================================
-; See https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP
+; See https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Exchanging-Device-and-Mailbox-State-Using-PJSIP/
 ; for more information.
 ;[inbound-publication]
 ;type=                     ; Must be of type 'inbound-publication'.
@@ -1579,7 +1579,7 @@
 
 ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_publish_asterisk
 ;==========================ASTERISK_PUBLICATION===============================
-; See https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP
+; See https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Exchanging-Device-and-Mailbox-State-Using-PJSIP/
 ; for more information.
 ;[asterisk-publication]
 ;type=asterisk-publication ; Must be of type 'asterisk-publication'.

configs/samples/pjsip_wizard.conf.sample

@@ -20,7 +20,7 @@
 
 ; Documentation
 ;
-; The official documentation is at http://wiki.asterisk.org
+; The official documentation is at https://docs.asterisk.org
 ; You can read the XML configuration help via Asterisk command line with
 ; "config show help res_pjsip_config_wizard", then you can drill down through
 ; the various sections and their options.

configs/samples/queues.conf.sample

@@ -286,6 +286,13 @@ monitor-type = MixMonitor
 ;
 ;periodic-announce-frequency=60
 ;
+; If given indicates the number of seconds after entering the queue the first
+; periodic announcement should be played.  The default (and historic) behavior
+; is to play the first periodic announcement at periodic-announce-frequency
+; seconds after entering the queue.
+;
+;periodic-announce-startdelay=10
+;
 ; Should the periodic announcements be played in a random order? Default is no.
 ;
 ;random-periodic-announce=no

configs/samples/res_config_odbc.conf.sample

@@ -0,0 +1,14 @@
+;
+; Sample configuration for res_config_odbc
+;
+; Most configuration occurs in the system ODBC configuration files,
+; res_odbc.conf, and extconfig.conf. You only need this file in the
+; event that you want to influence default sorting behavior.
+;
+
+[general]
+; When multiple rows are requested by realtime, res_config_odbc will add an
+; explicit ORDER BY clause to the generated SELECT statement. To prevent
+; that from occuring, set order_multi_row_results_by_initial_column to 'no'.
+;
+;order_multi_row_results_by_initial_column=no

configs/samples/res_pgsql.conf.sample

@@ -28,3 +28,9 @@ dbpass=password
 ; createchar  - Create char columns only
 ;
 requirements=warn
+
+; When multiple rows are requested by realtime, res_config_pgsql will add an
+; explicit ORDER BY clause to the generated SELECT statement. To prevent
+; that from occuring, set order_multi_row_results_by_initial_column to 'no'.
+;
+;order_multi_row_results_by_initial_column=no

configs/samples/sla.conf.sample

@@ -1,7 +1,7 @@
 ;
 ; Configuration for Shared Line Appearances (SLA).
 ;
-; See http://wiki.asterisk.org or doc/AST.pdf for more information.
+; See https://docs.asterisk.org for more information.
 ;
 
 ; ---- General Options ----------------
@@ -37,7 +37,7 @@
                             ;       DAHDI channels can be directly used.  IP trunks
                             ;       require some indirect configuration which is
                             ;       described in
-                            ; https://wiki.asterisk.org/wiki/display/AST/SLA+Trunk+Configuration
+                            ; https://docs.asterisk.org/Configuration/Applications/Shared-Line-Appearances-SLA/
 
 ;autocontext=line1          ; This supports automatic generation of the dialplan entries
                             ; if the autocontext option is used.  Each trunk should have
@@ -73,7 +73,7 @@
 ;type=trunk
 ;device=Local/disa@line4_outbound ; A Local channel in combination with the Disa
                                   ; application can be used to support IP trunks.
-                                  ; See https://wiki.asterisk.org/wiki/display/AST/SLA+Trunk+Configuration
+                                  ; See https://docs.asterisk.org/Configuration/Applications/Shared-Line-Appearances-SLA/
 ;autocontext=line4
 ; --------------------------------------
 

configs/samples/sorcery.conf.sample

@@ -76,3 +76,6 @@ test=memory
 
 ;[res_pjsip_publish_asterisk]
 ;asterisk-publication=realtime,ps_asterisk_publications
+
+;[res_stir_shaken]
+;tn=realtime,stir_tn

configs/samples/stir_shaken.conf.sample

@@ -1,103 +1,459 @@
-;
-; This file is used by the res_stir_shaken module to configure parameters
-; used for STIR/SHAKEN.
-;
-; There are 2 sides to STIR/SHAKEN: attestation and verification.
-;
-; Attestation is done on outgoing calls and makes use out of the certificate
-; objects. The cert located at path will be used to sign, and the cert
-; located at public_cert_url will be placed in the Identity header to let the
-; remote side know where to download the public cert from. These 2 certs must
-; match; that is, the cert located at public_cert_url must be the public cert
-; derived from the private cert located at path.
-;
-; Verification is done on incoming calls and doesn't rely on cert objects
-; defined in this file.
-;
-; The general section applies to all STIR/SHAKEN operations. However,
-; cache_max_size, curl_timeout, and signature_timeout only apply to the
-; verification side.
-;
-; It's important to note that downloaded certificates are stored in
-; <ast_config_AST_DATA_DIR>/keys/stir_shaken, which is usually
-; /etc/asterisk/keys/stir_shaken, but may be changed depending on where your
-; config directory is.
-;
-; Visit the wiki page:
-; https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN
-;
-; [general]
-;
-; File path to the certificate authority certificate
-;ca_file=/etc/asterisk/stir/ca.crt
-;
-; File path to a chain of trust
-;ca_path=/etc/asterisk/stir/ca
-;
-; Maximum size to use for caching public keys
-;cache_max_size=1000
-;
-; Maximum time (in seconds) to wait to CURL certificates
-;curl_timeout=2
-;
-; Amount of time (in seconds) a signature is valid for
-;signature_timeout=15
-;
-;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
-;
-; A certificate store is used to examine, and load all certificates found in a
-; given directory. When using this type the public key URL is generated based
-; upon the filename, and variable substitution.
-;[certificates]
-;
-; type must be "store"
-;type=store
-;
-; Path to a directory containing certificates
-;path=/etc/asterisk/stir
-;
-; URL to the public certificate(s). Must contain variable '${CERTIFICATE}' used for
-; substitution. '${CERTIFICATE}' will be replaced by the names of the files located
-; at path.
-; This will be put in the Identity header when signing.
-;public_cert_url=http://mycompany.com/${CERTIFICATE}.pem
-;
-;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
-;
-; Individual certificates are declared by using the certificate type.
-;[alice]
-;
-; type must be "certificate"
-;type=certificate
-;
-; File path to a certificate. This can be RSA or ECDSA, but eventually only ECDSA will be supported.
-;path=/etc/asterisk/stir/alice.pem
-;
-; URL to the public certificate. Must be of type X509 and be derived from the
-; certificate located at path.
-; This will be put in the identity header when signing.
-;public_cert_url=http://mycompany.com/alice.pem
-;
-; The caller ID number to match on
-;caller_id_number=1234567
-;
-; Must have an attestation of A, B, or C
-;attestation=C
-;
-;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
-;
-; Profiles can be defined here which can be referenced by channel drivers.
-;[my_profile]
-;
-; type must be "profile"
-;type=profile
-;
-; Set stir_shaken to 'attest', 'verify', or 'on', which is the default
-;stir_shaken=on
-;
-; You can specify an ACL that will be used strictly for the Identity header when downloading public certificates
-;acllist=myacllist
-;
-; You can also do permit / deny lines if you want (also supports IPv6)
+;--
+
+There are 4 object types used by the STIR/SHAKEN process...
+
+The "attestation" object sets the parameters for creating an Identity
+header which attests to the ownership of the caller id on outgoing
+INVITE requests.
+
+One or more "tn" objects that are used to create the outgoing Identity
+header.  Each object's "id" is a specific caller-id telephone number
+and the object contains the URL to the certificate that was used to
+attest to the ownership of the caller-id, the level (A,B,C) of the
+attestation you're making, and the private key the asterisk
+attestation service will use to sign the Identity header.  When
+an outgoing INVITE request is placed, the attestation service will
+look up the caller-id in the tn object list and if it's found, use
+the information in the object to create the Identity header.
+
+The "verification" object sets the parameters for verification
+of the Identity header and caller id on incoming INVITE requests.
+
+One or more "profile" objects that can be associated to channel
+driver endpoints (currently only chan_pjsip).  Profiles can set
+whether verification, attestation, both or neither should be
+performed on requests coming in to this endpoint or requests
+going out from this endpoint. Additionally they can override
+most of the attestation and verification options to make them
+specific to an endpoint.  When Asterisk loads the configs, it
+creates "effective profiles" or "eprofiles" on the fly that are
+the amalgamation of the attestation, verification and profile.
+You can see them in the CLI with "stir_shaken show eprofiles".
+
+NOTE: The "tn" object can be configured to source its data from a
+realtime database by configuring sorcery.conf and extconfig.conf.
+Both of those files have examples for "stir_tn".  There is also an
+Alembic script in the "config" section of contrib/ast-db-manage that
+will create the table.  Since there can be only one "verification"
+or "attestation" object, and will probably be only a few "profile"
+objects, those objects aren't realtime enabled.
+
+--;
+
+;--
+=======================================================================
+  Attestation Object Description
+=======================================================================
+The "attestation" object sets the parameters for creating an Identity
+header which attests to the ownership of the caller id on outgoing
+INVITE requests.
+
+All parameters except 'global_disable" may be overridden in a "profile"
+or "tn" object.
+
+Only one "attestation" object may exist.
+
+Parameters:
+
+-- global_disable -----------------------------------------------------
+If set, globally disables the attestation service.  No Identity headers
+will be added to any outgoing INVITE requests.
+
+Default: no
+
+-- private_key_file ---------------------------------------------------
+The path to a file containing the private key you received from the
+issuing authority.  The file must NOT be group or world readable or
+writable so make sure the user the asterisk process is running as is
+the owner.
+
+Default: none
+
+-- public_cert_url ----------------------------------------------------
+The URL to the certificate you received from the issueing authority.
+They may give you a URL to use or you may have to host the certificate
+yourself and provide your own URL here.
+
+Default: none
+
+WARNING:  Make absolutely sure the file that's made public doesn't
+accidentally include the privite key as well as the certificate.
+If you set "check_tn_cert_public_url" in the "attestation" section
+above, the tn will not be loaded and a "DANGER" message will be output
+on the asterisk console if the file does contain a private key.
+
+-- check_tn_cert_public_url -------------------------------------------
+Identity headers in outgoing requests must contain a URL that points
+to the certificate used to sign the header.  Setting this parameter
+tells Asterisk to actually try to retrieve the certificates indicated
+by "public_cert_url" parameters and fail loading that tn if the cert
+can't be retrieved or if its 'Not Valid Before" -> 'Not Valid After"
+date range doesn't include today.  This is a network intensive process
+so use with caution.
+
+Default: no
+
+-- attest_level -------------------------------------------------------
+The level of the attestation you're making.
+One of "A", "B", "C"
+
+Default: none
+
+-- send_mky -----------------------------------------------------------
+If set and an outgoing call uses DTLS, an "mky" Media Key grant will
+be added to the Identity header.  Although RFC8224/8225 require this,
+not many implementations support it so a remote verification service
+may fail to verify the signature.
+
+Default: no
+
+-----------------------------------------------------------------------
+Example "attestation" object:
+--;
+
+;[attestation]
+;global_disable = no
+;private_key_path = /var/lib/asterisk/keys/stir_shaken/tns/multi-tns-key.pem
+;public_cert_url = https://example.com/tncerts/multi-tns-cert.pem
+;attest_level = C
+
+;--
+=======================================================================
+  TN Object Description
+=======================================================================
+Each "tn" object contains the parameters needed to create the Identity
+header used to attest to the ownership of the caller-id on outgoing
+requests.  When an outgoing INVITE request is placed, the attestation
+service will look up the caller-id in this list and if it's found, use
+the information in the object to create the Identity header.
+The private key and certificate needed to sign the Identity header are
+usually provided to you by the telephone number issuing authority along
+with their certificate authority certificate.  You should give the CA
+certificate to any recipients who expect to receive calls from you
+although this has probably already been done by the issuing authority.
+
+The "id" of this object MUST be a canonicalized telephone number which
+starts with a country code.  The only valid characters are the numbers
+0-9, '#' and '*'.
+
+Parameters:
+
+-- type (required) ----------------------------------------------------
+Must be set to "tn"
+
+Default: none
+
+-- private_key_file ---------------------------------------------------
+The path to a file containing the private key you received from the
+issuing authority.  The file must NOT be group or world readable or
+writable so make sure the user the asterisk process is running as is
+the owner.
+
+Default: private_key_file from the profile or attestation objects.
+
+-- public_cert_url ----------------------------------------------------
+The URL to the certificate you received from the issueing authority.
+They may give you a URL to use or you may have to host the certificate
+yourself and provide your own URL here.
+
+Default: public_cert_url from the profile or attestation objects.
+
+WARNING:  Make absolutely sure the file that's made public doesn't
+accidentally include the privite key as well as the certificate.
+If you set "check_tn_cert_public_url" in the "attestation" section
+above, the tn will not be loaded and a "DANGER" message will be output
+on the asterisk console if the file does contain a private key.
+
+-- attest_level -------------------------------------------------------
+The level of the attestation you're making.
+One of "A", "B", "C"
+
+Default: attest_level from the profile or attestation objects.
+
+-----------------------------------------------------------------------
+Example "tn" object:
+--;
+
+;[18005551515]
+;type = tn
+;private_key_path = /var/lib/asterisk/keys/stir_shaken/tns/18005551515-key.pem
+;public_cert_url = https://example.com/tncerts/18005551515-cert.pem
+;attest_level = C
+
+;--
+=======================================================================
+  Verification Object Description
+=======================================================================
+The "verification" object sets the parameters for verification
+of the Identity header on incoming INVITE requests.
+
+All parameters except 'global_disable" may be overridden in a "profile"
+object.
+
+Only one "verification" object may exist.
+
+Parameters:
+
+-- global_disable -----------------------------------------------------
+If set, globally disables the verification service.
+
+Default: no
+
+-- load_system_certs---------------------------------------------------
+If set, loads the system Certificate Authority certificates
+(usually located in /etc/pki/CA) into the trust store used to
+validate the certificates in incoming requests.  This is not
+normally required as service providers will usually provide their
+CA certififcate to you separately.
+
+Default: no
+
+-- ca_file -----------------------------------------------------------
+Path to a single file containing a CA certificate or certificate chain
+to be used to validate the certificates in incoming requests.
+
+Default: none
+
+-- ca_path -----------------------------------------------------------
+Path to a directory containing one or more CA certificates to be used
+to validate the certificates in incoming requests.  The files in that
+directory must contain only one certificate each and the directory
+must be hashed using the OpenSSL 'c_rehash' utility.
+
+Default: none
+
+NOTE:  Both ca_file and ca_path can be specified but at least one
+MUST be.
+
+-- crl_file -----------------------------------------------------------
+Path to a single file containing a CA certificate revocation list
+to be used to validate the certificates in incoming requests.
+
+Default: none
+
+-- crl_path -----------------------------------------------------------
+Path to a directory containing one or more CA certificate revocation
+lists to be used to validate the certificates in incoming requests.
+The files in that directory must contain only one certificate each and
+the directory must be hashed using the OpenSSL 'c_rehash' utility.
+
+Default: none
+
+NOTE:  Neither crl_file nor crl_path are required.
+
+-- cert_cache_dir -----------------------------------------------------
+Incoming Identity headers will have a URL pointing to the certificate
+used to sign the header.  To prevent us from having to retrieve the
+certificate for every request, we maintain a cache of them in the
+'cert_cache_dir' specified.  The directory will be checked for
+existence and writability at startup.
+
+Default: <astvarlibdir>/keys/stir_shaken/cache
+
+-- curl_timeout -------------------------------------------------------
+The number of seconds we'll wait for a response when trying to retrieve
+the certificate specified in the incoming Identity header's "x5u"
+parameter.
+
+Default: 2
+
+-- max_cache_entry_age ------------------------------------------------
+Maximum age in seconds a certificate in the cache can reach before
+re-retrieving it.
+
+Default: 86400 (24 hours per ATIS-1000074)
+
+NOTE: If, when retrieving the URL specified by the "x5u" parameter,
+we receive a recognized caching directive in the HTTP response AND that
+directive indicates caching for MORE than the value set here, we'll use
+that time for the max_cache_entry_age.
+
+-- max_cache_size -----------------------------------------------------
+Maximum number of entries the cache can hold.
+Not presently implemented.
+
+-- max_iat_age --------------------------------------------------------
+The "iat" parameter in the Identity header indicates the time the
+sender actually created their attestation. If that is older than the
+current time by the number of seconds set here, the request will be
+considered "failed".
+
+Default: 15
+
+-- max_date_header_age ------------------------------------------------
+The sender MUST also send a SIP Date header in their request.  If we
+receive one that is older than the current time by the number of seconds
+set here, the request will be considered "failed".
+
+Default: 15
+
+-- failure_action -----------------------------------------------------
+Indicates what will happen to requests that have failed verification.
+Must be one of:
+- continue -
+  Continue processing the request.  You can use the STIR_SHAKEN
+  dialplan function to determine whether the request passed or failed
+  verification and take the action you deem appropriate.
+
+- reject_request -
+  Reject the request immediately using the SIP response codes
+  defined by RFC8224.
+
+- continue_return_reason -
+  Continue processing the request but, per RFC8224, send a SIP Reason
+  header back to the originator in the next provisional response
+  indicating the issue according to RFC8224.  You can use the
+  STIR_SHAKEN dialplan function to determine whether the request
+  passed or failed verification and take the action you deem
+  appropriate.
+
+Default: continue
+
+NOTE: If you select "continue" or "continue_return_reason", and,
+based on the results from the STIR_SHAKEN function, you determine you
+want to terminate the call, you can use the PJSIPHangup() dialplan
+application to reject the call using a STIR/SHAKEN-specific SIP
+response code.
+
+-- use_rfc9410_responses ----------------------------------------------
+If set, when sending Reason headers back to originators, the protocol
+header parameter will be set to "STIR" rather than "SIP".  This is a
+new protocol defined in RFC9410 and may not be supported by all
+participants.
+
+Default: no
+
+-- relax_x5u_port_scheme_restrictions ---------------------------------
+If set, the port and scheme restrictions imposed by ATIS-1000074
+section 5.3.1 that require the scheme to be "https" and the port to
+be 443 or 8443 are relaxed.  This will allow schemes like "http"
+and ports other than the two mentioned to appear in x5u URLs received
+in Identity headers.
+
+Default: no
+
+CAUTION: Setting this parameter could have serious security
+implications and should only be use for testing.
+
+-- relax_x5u_path_restrictions ----------------------------------------
+If set, the path restrictions imposed by ATIS-1000074 section 5.3.1
+that require the x5u URL to be rejected if it contains a query string,
+path parameters, fragment identifier or user/password are relaxed.
+
+Default: no
+
+CAUTION: Setting this parameter could have serious security
+implications and should only be use for testing.
+
+-- x5u_permit/x5u_deny ------------------------------------------------
+When set, the IP address of the host in a received Identity header x5u
+URL is checked against the acl created by this list of permit/deny
+parameters.  If the check fails, the x5u URL will be considered invalid
+and verification will fail.  This can prevent an attacker from sending
+you a request pretending to be a known originator with a mailcious
+certificate URL. (Server-side request forgery (SSRF)).
+See acl.conf.sample to see examples of how to specify the permit/deny
+parameters.
+
+Default:  Deny all "Special-Purpose" IP addresses described in RFC 6890.
+This includes the loopback addresses 127.0.0.0/8, private use networks such
+as 10.0.0/8, 172.16.0.0/12 and 192.168.0.0/16, and the link local network
+169.254.0.0/16 among others.
+
+CAUTION: Setting this parameter could have serious security
+implications and should only be use for testing.
+
+-- x5u_acl ------------------------------------------------------------
+Rather than providing individual permit/deny parameters, you can set
+the acllist parameter to an acl list predefined in acl.conf.
+
+Default: none
+
+CAUTION: Setting this parameter could have serious security
+implications and should only be use for testing.
+
+-----------------------------------------------------------------------
+Example "verification" object:
+--;
+
+;[verification]
+;global_disable = yes
+;load_system_certs = no
+;ca_path = /var/lib/asterisk/keys/stir_shaken/verification_ca
+;cert_cache_dir = /var/lib/asterisk/keys/stir_shaken/verification_cache
+;failure_action = reject_request
+;curl_timeout=5
+;max_iat_age=60
+;max_date_header_age=60
+;max_cache_entry_age = 300
+; For internal testing
+;x5u_deny=0.0.0.0/0.0.0.0
+;x5u_permit=127.0.0.0/8
+;x5u_permit=192.168.100.0/24
+;relax_x5u_port_scheme_restrictions = yes
+;relax_x5u_path_restrictions = yes
+
+;--
+=======================================================================
+  Profile Object Description
+=======================================================================
+A "profile" object can be associated to channel driver endpoint
+(currently only chan_pjsip) and can set verification and attestation
+parameters specific to endpoints using this profile.  If you have
+multiple upstream providers, this is the place to set parameters
+specific to them.
+
+The "id" of this object is arbitrary and you'd specify it in the
+"stir_shaken_profile" parameter of the endpoint.
+
+Parameters:
+
+-- type (required) ----------------------------------------------------
+Must be set to "profile"
+
+Default: none
+
+-- endpoint_behhavior--------------------------------------------------
+Actions to be performed for endpoints referencing this profile.
+Must be one of:
+- off -
+  Don't do any STIR/SHAKEN processing.
+- attest -
+  Attest on outgoing calls.
+- verify
+  Verify incoming calls.
+- on -
+  Attest outgoing calls and verify incoming calls.
+Default: off
+
+All of the "verification" parameters defined above can be set on a profile
+with the exception of 'global_disable'.
+
+All of the "attestation" parameters defined above can be set on a profile
+with the exception of 'global_disable'.
+
+When Asterisk loads the configs, it creates "effective profiles" or
+"eprofiles" on the fly that are the amalgamation of the attestation,
+verification and profile. You can see them in the CLI with
+"stir_shaken show eprofiles".
+
+-----------------------------------------------------------------------
+Example "profile" object:
+--;
+
+;[myprofile]
+;type = profile
+;endpoint_behavior = verify
+;failure_action = continue_return_reason
+;x5u_acl = myacllist
+
+;In pjsip.conf...
+;[myendpoint]
+;type = endpoint
+;stir_shaken_profile = myprofile
+
+;In acl.conf...
+;[myacllist]
 ;permit=0.0.0.0/0.0.0.0
-;deny=127.0.0.1
+;deny=10.24.20.171
+

configs/samples/unistim.conf.sample

@@ -5,7 +5,7 @@
 [general]
 port=5000                    ; UDP port
 ;
-; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
+; See https://docs.asterisk.org/Configuration/Channel-Drivers/IP-Quality-of-Service for a description of these parameters.
 ;tos=cs3                ; Sets TOS for signaling packets.
 ;tos_audio=ef           ; Sets TOS for RTP audio packets.
 ;cos=3                  ; Sets 802.1p priority for signaling packets.

configs/samples/voicemail.conf.sample

@@ -293,7 +293,8 @@ sendvoicemail=yes ; Allow the user to compose and send a voicemail while inside
 			;     if not listed, calling the sender back will not be permitted
 ; exitcontext=fromvm    ; Context to go to on user exit such as * or 0
                         ;     The default is the current context.
-; review=yes 		; Allow sender to review/rerecord their message before saving it [OFF by default
+; review=yes 		; Allow sender to review/rerecord their message before saving it [OFF by default]
+; leaveurgent=yes   ; Allow senders to leave messages that are marked as 'Urgent' [ON by default]
 ; operator=yes      ; Allow sender to hit 0 before/after/during leaving a voicemail to
                     ; reach an operator.  This option REQUIRES an 'o' extension in the
                     ; same context (or in exitcontext, if set), as that is where the

configure.ac

@@ -1,6 +1,6 @@
 AC_PREREQ(2.60a)
 
-AC_INIT([asterisk], [master], [https://github.com/asterisk/asterisk/issues])
+AC_INIT([asterisk], [21], [https://github.com/asterisk/asterisk/issues])
 
 # cross-compile macros
 AC_CANONICAL_BUILD
@@ -463,6 +463,15 @@ if test "${with_pjproject}" = "no" || test "${with_pjproject}" = "n" ; then
 	PJPROJECT_BUNDLED=no
 fi
 
+LIBJWT_BUNDLED=no
+AC_ARG_WITH([libjwt-bundled],
+	[AS_HELP_STRING([--with-libjwt-bundled],
+		[Use bundled libjwt library])],
+	[case "${withval}" in
+		y|ye|yes) LIBJWT_BUNDLED=yes ;;
+		*) LIBJWT_BUNDLED=no ;;
+	esac])
+
 #
 # OpenSSL stuff has to be done here because we want to pass
 # any resulting CFLAGS and LDFLAGS to the bundled pjproject
@@ -553,6 +562,7 @@ AST_EXT_LIB_SETUP([LDAP], [OpenLDAP], [ldap])
 AST_LIBCURL_CHECK_CONFIG([], [7.10.1])
 AST_EXT_LIB_SETUP([LIBEDIT], [NetBSD Editline library], [libedit])
 AST_EXT_LIB_SETUP_OPTIONAL([LIBEDIT_IS_UNICODE], [Libedit compiled for unicode], [LIBEDIT], [libedit])
+AST_EXT_LIB_SETUP([LIBJWT], [LIBJWT], [libjwt])
 AST_EXT_LIB_SETUP([LIBXML2], [LibXML2], [libxml2])
 AST_EXT_LIB_SETUP([LIBXSLT], [LibXSLT], [libxslt])
 AST_EXT_LIB_SETUP_OPTIONAL([LIBXSLT_CLEANUP], [LibXSLT Library Cleanup Function], [LIBXSLT], [libxslt])
@@ -736,6 +746,14 @@ else
 	PBX_JANSSON=1
 fi
 
+source ./third-party/versions.mak
+# Find required JWT support if bundled is not enabled.
+if test "$LIBJWT_BUNDLED" = "no" ; then
+	AST_PKG_CONFIG_CHECK([LIBJWT], [libjwt >= $LIBJWT_VERSION])
+else
+	PBX_LIBJWT=1
+fi
+
 # See if clock_gettime is in librt
 AST_EXT_LIB_CHECK([RT], [rt], [clock_gettime], [])
 
@@ -1154,9 +1172,6 @@ AC_LINK_IFELSE(
 #)
 #fi
 
-# for FreeBSD thr_self
-AC_CHECK_HEADERS([sys/thr.h])
-
 AC_MSG_CHECKING(for compiler sync operations)
 AC_LINK_IFELSE(
 [AC_LANG_PROGRAM([], [int foo1; int foo2 = __sync_fetch_and_add(&foo1, 1);])],
@@ -1714,7 +1729,7 @@ if test "${USE_ILBC}" != "no"; then
       ILBC_INTERNAL="no"
    fi
    if test "${ILBC_SYSTEM}" = "yes"; then
-      AST_PKG_CONFIG_CHECK(ILBC, libilbc)
+      AST_PKG_CONFIG_CHECK(ILBC, libilbc < 3)
       if test "$PBX_ILBC" = "1"; then
 	 ILBC_INTERNAL="no"
       fi

contrib/ast-db-manage/.gitignore

@@ -0,0 +1 @@
+*.ini

contrib/ast-db-manage/config/versions/24c12d8e9014_add_user_agent_header_to_ps_.py

@@ -0,0 +1,22 @@
+"""add user-agent-header to ps_registrations
+
+Revision ID: 24c12d8e9014
+Revises: 37a5332640e2
+Create Date: 2024-01-05 14:14:47.510917
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '24c12d8e9014'
+down_revision = '37a5332640e2'
+
+from alembic import op
+import sqlalchemy as sa
+
+
+def upgrade():
+    op.add_column('ps_registrations', sa.Column('user_agent', sa.String(255)))
+
+
+def downgrade():
+    op.drop_column('ps_registrations', 'user_agent')

contrib/ast-db-manage/config/versions/37a5332640e2_update_pjsip_tls_method_list.py

@@ -0,0 +1,58 @@
+"""update pjsip tls method list
+
+Revision ID: 37a5332640e2
+Revises: dac2b4c328b8
+Create Date: 2023-11-14 18:02:18.857452
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '37a5332640e2'
+down_revision = 'dac2b4c328b8'
+
+from alembic import op
+from sqlalchemy.dialects.postgresql import ENUM
+import sqlalchemy as sa
+
+PJSIP_TRANSPORT_METHOD_OLD_NAME = 'pjsip_transport_method_values'
+PJSIP_TRANSPORT_METHOD_NEW_NAME = 'pjsip_transport_method_values_v2'
+
+PJSIP_TRANSPORT_METHOD_OLD_VALUES = ['default', 'unspecified', 'tlsv1', 'sslv2',
+                                     'sslv3', 'sslv23']
+PJSIP_TRANSPORT_METHOD_NEW_VALUES = ['default', 'unspecified',
+                                     'tlsv1', 'tlsv1_1', 'tlsv1_2', 'tlsv1_3',
+                                     'sslv2', 'sslv23', 'sslv3']
+
+PJSIP_TRANSPORT_METHOD_OLD_TYPE = sa.Enum(*PJSIP_TRANSPORT_METHOD_OLD_VALUES,
+                                          name=PJSIP_TRANSPORT_METHOD_OLD_NAME)
+PJSIP_TRANSPORT_METHOD_NEW_TYPE = sa.Enum(*PJSIP_TRANSPORT_METHOD_NEW_VALUES,
+                                          name=PJSIP_TRANSPORT_METHOD_NEW_NAME)
+def upgrade():
+    if op.get_context().bind.dialect.name == 'postgresql':
+        enum = PJSIP_TRANSPORT_METHOD_NEW_TYPE
+        enum.create(op.get_bind(), checkfirst=False)
+
+    op.alter_column('ps_transports', 'method',
+                    type_=PJSIP_TRANSPORT_METHOD_NEW_TYPE,
+                    existing_type=PJSIP_TRANSPORT_METHOD_OLD_TYPE,
+                    postgresql_using='method::text::' + PJSIP_TRANSPORT_METHOD_NEW_NAME)
+
+    if op.get_context().bind.dialect.name == 'postgresql':
+        ENUM(name=PJSIP_TRANSPORT_METHOD_OLD_NAME).drop(op.get_bind(), checkfirst=False)
+
+def downgrade():
+    # First we need to ensure that columns are not using the enum values
+    # that are going away.
+    op.execute("UPDATE ps_transports SET method = 'tlsv1' WHERE method IN ('tlsv1_1', 'tlsv1_2', 'tlsv1_3')")
+
+    if op.get_context().bind.dialect.name == 'postgresql':
+        enum = PJSIP_TRANSPORT_METHOD_OLD_TYPE
+        enum.create(op.get_bind(), checkfirst=False)
+
+    op.alter_column('ps_transports', 'method',
+                    type_=PJSIP_TRANSPORT_METHOD_OLD_TYPE,
+                    existing_type=PJSIP_TRANSPORT_METHOD_NEW_TYPE,
+                    postgresql_using='method::text::' + PJSIP_TRANSPORT_METHOD_OLD_NAME)
+
+    if op.get_context().bind.dialect.name == 'postgresql':
+        ENUM(name=PJSIP_TRANSPORT_METHOD_NEW_NAME).drop(op.get_bind(), checkfirst=False)

contrib/ast-db-manage/config/versions/539f68bede2c_add_peer_supported_to_100rel.py

@@ -33,9 +33,9 @@ def upgrade():
         enum = ENUM(*NEW_ENUM, name='pjsip_100rel_values_v2')
         enum.create(op.get_bind(), checkfirst=False)
 
-        op.execute('ALTER TABLE ps_endpoints ALTER COLUMN 100rel TYPE'
+        op.execute('ALTER TABLE ps_endpoints ALTER COLUMN "100rel" TYPE'
                    ' pjsip_100rel_values_v2 USING'
-                   ' 100rel::text::pjsip_100rel_values_v2')
+                   ' "100rel"::text::pjsip_100rel_values_v2')
 
         ENUM(name="pjsip_100rel_values").drop(op.get_bind(), checkfirst=False)
 
@@ -50,8 +50,8 @@ def downgrade():
         enum = ENUM(*OLD_ENUM, name='pjsip_100rel_values')
         enum.create(op.get_bind(), checkfirst=False)
 
-        op.execute('ALTER TABLE ps_endpoints ALTER COLUMN 100rel TYPE'
+        op.execute('ALTER TABLE ps_endpoints ALTER COLUMN "100rel" TYPE'
                    ' pjsip_100rel_values USING'
-                   ' 100rel::text::pjsip_100rel_values')
+                   ' "100rel"::text::pjsip_100rel_values')
 
         ENUM(name="pjsip_100rel_values_v2").drop(op.get_bind(), checkfirst=False)

contrib/ast-db-manage/config/versions/74dc751dfe8e_more_permission_boolean_columns.py

@@ -0,0 +1,158 @@
+"""more permission boolean columns
+
+Revision ID: 74dc751dfe8e
+Revises: bd335bae5d33
+Create Date: 2024-02-27 15:31:09.458313
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '74dc751dfe8e'
+down_revision = 'bd335bae5d33'
+
+import itertools
+import operator
+
+from alembic import op
+import sqlalchemy as sa
+from sqlalchemy import case, cast, or_, text
+from sqlalchemy.dialects.postgresql import ENUM
+from sqlalchemy.sql import table, column
+
+COLUMNS = [ ('ps_aors', 'authenticate_qualify'),
+            ('ps_aors', 'remove_existing'),
+            ('ps_aors', 'remove_unavailable'),
+            ('ps_aors', 'support_path'),
+            ('ps_asterisk_publications', 'device_state'),
+            ('ps_asterisk_publications', 'mailbox_state'),
+            ('ps_contacts', 'authenticate_qualify'),
+            ('ps_contacts', 'prune_on_boot'),
+            ('ps_endpoint_id_ips', 'srv_lookups'),
+            ('ps_endpoints', '100rel'),
+            ('ps_endpoints', 'accept_multiple_sdp_answers'),
+            ('ps_endpoints', 'aggregate_mwi'),
+            ('ps_endpoints', 'allow_overlap'),
+            ('ps_endpoints', 'allow_subscribe'),
+            ('ps_endpoints', 'allow_transfer'),
+            ('ps_endpoints', 'allow_unauthenticated_options'),
+            ('ps_endpoints', 'asymmetric_rtp_codec'),
+            ('ps_endpoints', 'bind_rtp_to_media_address'),
+            ('ps_endpoints', 'bundle'),
+            ('ps_endpoints', 'direct_media'),
+            ('ps_endpoints', 'disable_direct_media_on_nat'),
+            ('ps_endpoints', 'dtls_auto_generate_cert'),
+            ('ps_endpoints', 'fax_detect'),
+            ('ps_endpoints', 'follow_early_media_fork'),
+            ('ps_endpoints', 'force_avp'),
+            ('ps_endpoints', 'force_rport'),
+            ('ps_endpoints', 'g726_non_standard'),
+            ('ps_endpoints', 'ice_support'),
+            ('ps_endpoints', 'ignore_183_without_sdp'),
+            ('ps_endpoints', 'inband_progress'),
+            ('ps_endpoints', 'media_encryption_optimistic'),
+            ('ps_endpoints', 'media_use_received_transport'),
+            ('ps_endpoints', 'moh_passthrough'),
+            ('ps_endpoints', 'mwi_subscribe_replaces_unsolicited'),
+            ('ps_endpoints', 'notify_early_inuse_ringing'),
+            ('ps_endpoints', 'one_touch_recording'),
+            ('ps_endpoints', 'preferred_codec_only'),
+            ('ps_endpoints', 'refer_blind_progress'),
+            ('ps_endpoints', 'rewrite_contact'),
+            ('ps_endpoints', 'rpid_immediate'),
+            ('ps_endpoints', 'rtcp_mux'),
+            ('ps_endpoints', 'rtp_ipv6'),
+            ('ps_endpoints', 'rtp_symmetric'),
+            ('ps_endpoints', 'send_aoc'),
+            ('ps_endpoints', 'send_connected_line'),
+            ('ps_endpoints', 'send_diversion'),
+            ('ps_endpoints', 'send_history_info'),
+            ('ps_endpoints', 'send_pai'),
+            ('ps_endpoints', 'send_rpid'),
+            ('ps_endpoints', 'srtp_tag_32'),
+            ('ps_endpoints', 'stir_shaken'),
+            ('ps_endpoints', 'suppress_q850_reason_headers'),
+            ('ps_endpoints', 't38_bind_udptl_to_media_address'),
+            ('ps_endpoints', 't38_udptl'),
+            ('ps_endpoints', 't38_udptl_ipv6'),
+            ('ps_endpoints', 't38_udptl_nat'),
+            ('ps_endpoints', 'timers'),
+            ('ps_endpoints', 'trust_connected_line'),
+            ('ps_endpoints', 'trust_id_inbound'),
+            ('ps_endpoints', 'trust_id_outbound'),
+            ('ps_endpoints', 'use_avpf'),
+            ('ps_endpoints', 'use_ptime'),
+            ('ps_endpoints', 'user_eq_phone'),
+            ('ps_endpoints', 'webrtc'),
+            ('ps_globals', 'all_codecs_on_empty_reinvite'),
+            ('ps_globals', 'allow_sending_180_after_183'),
+            ('ps_globals', 'disable_multi_domain'),
+            ('ps_globals', 'ignore_uri_user_options'),
+            ('ps_globals', 'mwi_disable_initial_unsolicited'),
+            ('ps_globals', 'norefersub'),
+            ('ps_globals', 'send_contact_status_on_update_registration'),
+            ('ps_globals', 'use_callerid_contact'),
+            ('ps_outbound_publishes', 'multi_user'),
+            ('ps_registrations', 'auth_rejection_permanent'),
+            ('ps_registrations', 'line'),
+            ('ps_registrations', 'support_path'),
+            ('ps_resource_list', 'full_state'),
+            ('ps_resource_list', 'resource_display_name'),
+            ('ps_subscription_persistence', 'prune_on_boot'),
+            ('ps_systems', 'accept_multiple_sdp_answers'),
+            ('ps_systems', 'compact_headers'),
+            ('ps_systems', 'disable_rport'),
+            ('ps_systems', 'disable_tcp_switch'),
+            ('ps_systems', 'follow_early_media_fork'),
+            ('ps_transports', 'allow_reload'),
+            ('ps_transports', 'allow_wildcard_certs'),
+            ('ps_transports', 'require_client_cert'),
+            ('ps_transports', 'symmetric_transport'),
+            ('ps_transports', 'verify_client'),
+            ('ps_transports', 'verify_server') ]
+
+YESNO_NAME = 'yesno_values'
+YESNO_VALUES = ['yes', 'no']
+
+AST_BOOL_NAME = 'ast_bool_values'
+AST_BOOL_VALUES = [ '0', '1',
+                    'off', 'on',
+                    'false', 'true',
+                    'no', 'yes' ]
+
+yesno_values = ENUM(*YESNO_VALUES, name=YESNO_NAME, create_type=False)
+ast_bool_values = ENUM(*AST_BOOL_VALUES, name=AST_BOOL_NAME, create_type=False)
+
+def upgrade():
+    for table_name, column_list in itertools.groupby(COLUMNS, operator.itemgetter(0)):
+        with op.batch_alter_table(table_name) as batch_op:
+            for _, column_name in column_list:
+                batch_op.alter_column(column_name,
+                                      type_=ast_bool_values,
+                                      existing_type=yesno_values,
+                                      postgresql_using='"{}"::text::{}'.format(column_name, AST_BOOL_NAME))
+
+def downgrade():
+    for table_name, column_list in itertools.groupby(COLUMNS, operator.itemgetter(0)):
+        subject = table(table_name)
+        values_exprs = {}
+        for _, column_name in column_list:
+            subject.append_column(column(column_name))
+            values_exprs[column_name] = cast(
+                case((or_(subject.c[column_name] == text("'yes'"),
+                          subject.c[column_name] == text("'1'"),
+                          subject.c[column_name] == text("'on'"),
+                          subject.c[column_name] == text("'true'")), text("'yes'")),
+                     else_=text("'no'")),
+                ast_bool_values)
+
+        op.execute(
+            subject.update().values(values_exprs)
+        )
+
+    for table_name, column_list in itertools.groupby(COLUMNS, operator.itemgetter(0)):
+        with op.batch_alter_table(table_name) as batch_op:
+            for _, column_name in column_list:
+                batch_op.alter_column(column_name,
+                                      type_=yesno_values,
+                                      existing_type=ast_bool_values,
+                                      postgresql_using='"{}"::text::{}'.format(column_name, YESNO_NAME))

contrib/ast-db-manage/config/versions/bd335bae5d33_create_stir_shaken_tn_table.py

@@ -0,0 +1,35 @@
+"""Create STIR/SHAKEN TN table
+
+Revision ID: bd335bae5d33
+Revises: 24c12d8e9014
+Create Date: 2024-01-09 12:17:47.353533
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = 'bd335bae5d33'
+down_revision = '24c12d8e9014'
+
+from alembic import op
+import sqlalchemy as sa
+from sqlalchemy.dialects.postgresql import ENUM
+
+AST_BOOL_NAME = 'ast_bool_values'
+AST_BOOL_VALUES = [ '0', '1',
+                    'off', 'on',
+                    'false', 'true',
+                    'no', 'yes' ]
+
+def upgrade():
+    ast_bool_values = ENUM(*AST_BOOL_VALUES, name=AST_BOOL_NAME, create_type=False)
+    op.create_table(
+        'stir_tn',
+        sa.Column('id', sa.String(80), nullable=False, primary_key=True),
+        sa.Column('private_key_file', sa.String(1024), nullable=True),
+        sa.Column('public_cert_url', sa.String(1024), nullable=True),
+        sa.Column('attest_level', sa.String(1), nullable=True),
+        sa.Column('send_mky', ast_bool_values)
+    )
+
+def downgrade():
+    op.drop_table('stir_tn')

contrib/ast-db-manage/config/versions/dac2b4c328b8_incease_pjsip_id_length.py

@@ -0,0 +1,83 @@
+"""increase pjsip id length
+
+Revision ID: dac2b4c328b8
+Revises: f5b0e7427449
+Create Date: 2023-09-23 02:15:24.270526
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = 'dac2b4c328b8'
+down_revision = 'f5b0e7427449'
+
+from alembic import op
+import sqlalchemy as sa
+
+
+def upgrade():
+    op.alter_column('ps_aors', 'id', type_=sa.String(255))
+    op.alter_column('ps_aors', 'outbound_proxy', type_=sa.String(255))
+
+    op.alter_column('ps_auths', 'id', type_=sa.String(255))
+    op.alter_column('ps_auths', 'realm', type_=sa.String(255))
+
+    op.alter_column('ps_contacts', 'outbound_proxy', type_=sa.String(255))
+    op.alter_column('ps_contacts', 'endpoint', type_=sa.String(255))
+
+    op.alter_column('ps_domain_aliases', 'id', type_=sa.String(255))
+    op.alter_column('ps_domain_aliases', 'domain', type_=sa.String(255))
+
+    op.alter_column('ps_endpoint_id_ips', 'id', type_=sa.String(255))
+    op.alter_column('ps_endpoint_id_ips', 'endpoint', type_=sa.String(255))
+
+    op.alter_column('ps_endpoints', 'id', type_=sa.String(255))
+    op.alter_column('ps_endpoints', 'aors', type_=sa.String(2048))
+    op.alter_column('ps_endpoints', 'auth', type_=sa.String(255))
+    op.alter_column('ps_endpoints', 'outbound_auth', type_=sa.String(255))
+    op.alter_column('ps_endpoints', 'outbound_proxy', type_=sa.String(255))
+
+    op.alter_column('ps_inbound_publications', 'id', type_=sa.String(255))
+    op.alter_column('ps_inbound_publications', 'endpoint', type_=sa.String(255))
+
+    op.alter_column('ps_outbound_publishes', 'id', type_=sa.String(255))
+    op.alter_column('ps_outbound_publishes', 'outbound_auth', type_=sa.String(255))
+
+    op.alter_column('ps_registrations', 'id', type_=sa.String(255))
+    op.alter_column('ps_registrations', 'outbound_auth', type_=sa.String(255))
+    op.alter_column('ps_registrations', 'outbound_proxy', type_=sa.String(255))
+    op.alter_column('ps_registrations', 'endpoint', type_=sa.String(255))
+
+
+def downgrade():
+    op.alter_column('ps_aors', 'id', type_=sa.String(40))
+    op.alter_column('ps_aors', 'outbound_proxy', type_=sa.String(40))
+
+    op.alter_column('ps_auths', 'id', type_=sa.String(40))
+    op.alter_column('ps_auths', 'realm', type_=sa.String(40))
+
+    op.alter_column('ps_contacts', 'outbound_proxy', type_=sa.String(40))
+    op.alter_column('ps_contacts', 'endpoint', type_=sa.String(40))
+
+    op.alter_column('ps_domain_aliases', 'id', type_=sa.String(40))
+    op.alter_column('ps_domain_aliases', 'domain', type_=sa.String(40))
+
+    op.alter_column('ps_endpoint_id_ips', 'id', type_=sa.String(40))
+    op.alter_column('ps_endpoint_id_ips', 'endpoint', type_=sa.String(40))
+
+    op.alter_column('ps_endpoints', 'id', type_=sa.String(40))
+    op.alter_column('ps_endpoints', 'aors', type_=sa.String(200))
+    op.alter_column('ps_endpoints', 'auth', type_=sa.String(40))
+    op.alter_column('ps_endpoints', 'outbound_auth', type_=sa.String(40))
+    op.alter_column('ps_endpoints', 'outbound_proxy', type_=sa.String(40))
+
+    op.alter_column('ps_inbound_publications', 'id', type_=sa.String(40))
+    op.alter_column('ps_inbound_publications', 'endpoint', type_=sa.String(40))
+
+    op.alter_column('ps_outbound_publishes', 'id', type_=sa.String(40))
+    op.alter_column('ps_outbound_publishes', 'outbound_auth', type_=sa.String(40))
+
+    op.alter_column('ps_registrations', 'id', type_=sa.String(40))
+    op.alter_column('ps_registrations', 'outbound_auth', type_=sa.String(40))
+    op.alter_column('ps_registrations', 'outbound_proxy', type_=sa.String(40))
+    op.alter_column('ps_registrations', 'endpoint', type_=sa.String(40))
+

contrib/ast-db-manage/voicemail/versions/1c55c341360f_remove_macrocontext_field.py

@@ -0,0 +1,22 @@
+"""Remove macrocontext field
+
+Revision ID: 1c55c341360f
+Revises: 39428242f7f5
+Create Date: 2024-01-09 15:01:39.698918
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '1c55c341360f'
+down_revision = '39428242f7f5'
+
+from alembic import op
+import sqlalchemy as sa
+
+
+def upgrade():
+    op.drop_column('voicemail_messages', 'macrocontext')
+
+
+def downgrade():
+    op.add_column('voicemail_messages', sa.Column('macrocontext', sa.String(80)))

contrib/realtime/mysql/mysql_cdr.sql

@@ -0,0 +1,41 @@
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> 210693f3123d
+
+CREATE TABLE cdr (
+    accountcode VARCHAR(20), 
+    src VARCHAR(80), 
+    dst VARCHAR(80), 
+    dcontext VARCHAR(80), 
+    clid VARCHAR(80), 
+    channel VARCHAR(80), 
+    dstchannel VARCHAR(80), 
+    lastapp VARCHAR(80), 
+    lastdata VARCHAR(80), 
+    start DATETIME, 
+    answer DATETIME, 
+    end DATETIME, 
+    duration INTEGER, 
+    billsec INTEGER, 
+    disposition VARCHAR(45), 
+    amaflags VARCHAR(45), 
+    userfield VARCHAR(256), 
+    uniqueid VARCHAR(150), 
+    linkedid VARCHAR(150), 
+    peeraccount VARCHAR(20), 
+    sequence INTEGER
+);
+
+INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
+
+-- Running upgrade 210693f3123d -> 54cde9847798
+
+ALTER TABLE cdr MODIFY accountcode VARCHAR(80) NULL;
+
+ALTER TABLE cdr MODIFY peeraccount VARCHAR(80) NULL;
+
+UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';
+

contrib/realtime/mysql/mysql_queue_log.sql

@@ -0,0 +1,29 @@
+BEGIN;
+
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> 4105ee839f58
+
+CREATE TABLE queue_log (
+    id BIGSERIAL NOT NULL, 
+    time TIMESTAMP WITHOUT TIME ZONE, 
+    callid VARCHAR(80), 
+    queuename VARCHAR(256), 
+    agent VARCHAR(80), 
+    event VARCHAR(32), 
+    data1 VARCHAR(100), 
+    data2 VARCHAR(100), 
+    data3 VARCHAR(100), 
+    data4 VARCHAR(100), 
+    data5 VARCHAR(100), 
+    PRIMARY KEY (id), 
+    UNIQUE (id)
+);
+
+INSERT INTO alembic_version (version_num) VALUES ('4105ee839f58');
+
+COMMIT;
+

contrib/realtime/mysql/mysql_voicemail.sql

@@ -0,0 +1,41 @@
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> a2e9769475e
+
+CREATE TABLE voicemail_messages (
+    dir VARCHAR(255) NOT NULL, 
+    msgnum INTEGER NOT NULL, 
+    context VARCHAR(80), 
+    macrocontext VARCHAR(80), 
+    callerid VARCHAR(80), 
+    origtime INTEGER, 
+    duration INTEGER, 
+    recording BLOB, 
+    flag VARCHAR(30), 
+    category VARCHAR(30), 
+    mailboxuser VARCHAR(30), 
+    mailboxcontext VARCHAR(30), 
+    msg_id VARCHAR(40)
+);
+
+ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
+
+CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
+
+INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
+
+-- Running upgrade a2e9769475e -> 39428242f7f5
+
+ALTER TABLE voicemail_messages MODIFY recording BLOB(4294967295) NULL;
+
+UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
+
+-- Running upgrade 39428242f7f5 -> 1c55c341360f
+
+ALTER TABLE voicemail_messages DROP COLUMN macrocontext;
+
+UPDATE alembic_version SET version_num='1c55c341360f' WHERE alembic_version.version_num = '39428242f7f5';
+

contrib/realtime/postgresql/postgresql_cdr.sql

@@ -0,0 +1,45 @@
+BEGIN;
+
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> 210693f3123d
+
+CREATE TABLE cdr (
+    accountcode VARCHAR(20), 
+    src VARCHAR(80), 
+    dst VARCHAR(80), 
+    dcontext VARCHAR(80), 
+    clid VARCHAR(80), 
+    channel VARCHAR(80), 
+    dstchannel VARCHAR(80), 
+    lastapp VARCHAR(80), 
+    lastdata VARCHAR(80), 
+    start TIMESTAMP WITHOUT TIME ZONE, 
+    answer TIMESTAMP WITHOUT TIME ZONE, 
+    "end" TIMESTAMP WITHOUT TIME ZONE, 
+    duration INTEGER, 
+    billsec INTEGER, 
+    disposition VARCHAR(45), 
+    amaflags VARCHAR(45), 
+    userfield VARCHAR(256), 
+    uniqueid VARCHAR(150), 
+    linkedid VARCHAR(150), 
+    peeraccount VARCHAR(20), 
+    sequence INTEGER
+);
+
+INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
+
+-- Running upgrade 210693f3123d -> 54cde9847798
+
+ALTER TABLE cdr ALTER COLUMN accountcode TYPE VARCHAR(80);
+
+ALTER TABLE cdr ALTER COLUMN peeraccount TYPE VARCHAR(80);
+
+UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';
+
+COMMIT;
+