<p>Kevin Harwell has uploaded this change for <strong>review</strong>.</p><p><a href="https://gerrit.asterisk.org/8045">View Change</a></p><pre style="font-family: monospace,monospace; white-space: pre-wrap;">Initial commit for new master testsuite branch<br><br>This branch will become the new master testsuite branch once the branches are<br>deemed ready (the current master branch will be suitably renamed).<br><br>Since the the Asterisk master branch is what will be come Asterisk 16.0.0 this<br>patch removes all tests that are not supported by the current Asterisk master<br>branch. Subsequently, this patch removes all tests that have a minversion<br>greater than 16.0.0 or a maxversion less than 16.0.0.<br><br>Change-Id: I93793406961fed649f89a46c775932004be07769<br>---<br>D tests/channels/SIP/redirecting_reason/v13.0.0/test-config.yaml<br>D tests/channels/pjsip/refer_send_to_vm/v12.2.0/test-config.yaml<br>2 files changed, 0 insertions(+), 121 deletions(-)<br><br></pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/45/8045/1</pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">diff --git a/tests/channels/SIP/redirecting_reason/v13.0.0/test-config.yaml b/tests/channels/SIP/redirecting_reason/v13.0.0/test-config.yaml<br>deleted file mode 100644<br>index 15bb36a..0000000<br>--- a/tests/channels/SIP/redirecting_reason/v13.0.0/test-config.yaml<br>+++ /dev/null<br>@@ -1,67 +0,0 @@<br>-testinfo:<br>- summary: 'Test that custom SIP redirecting reasons are accessible from the dialplan'<br>- description: |<br>- 'This test performs three calls.<br>-<br>- On the first call, an INVITE with a Diversion header arrives into Asterisk. We<br>- ensure that from the dialplan, the redirecting reason is what we expect it to be.<br>-<br>- On the second call, we place an outgoing call to a UAS that responds with a 302. In<br>- that 302, there is a Diversion header with a reason parameter. We again check in the<br>- dialplan to ensure that the redirecting reason has been set to this value.<br>-<br>- On the third call, we place an outgoing call to a UAS that responds with a 480, and<br>- with custom status text. We then ensure that the custom status text appears as the<br>- redirecting reason in the dialplan.'<br>-<br>-test-modules:<br>- test-object:<br>- config-section: 'calls'<br>- typename: 'sipp.SIPpTestCase'<br>- modules:<br>- -<br>- config-section: 'ami-config'<br>- typename: 'ami.AMIEventModule'<br>-<br>-calls:<br>- fail-on-any: False<br>- test-iterations:<br>- -<br>- scenarios:<br>- - { 'key-args': { 'scenario': 'uac-diversion.xml', '-s': 'test_diversion'} }<br>- -<br>- scenarios:<br>- - { 'key-args': { 'scenario': 'uas-redirect.xml', '-p': '5062'},<br>- 'ordered-args': ['-key', 'redir_target', 'test_diversion']}<br>- - { 'key-args': { 'scenario': 'uac-no-hangup.xml', '-d': '2000', '-s': 'diverter', '-p': '5061'} }<br>- -<br>- scenarios:<br>- - { 'key-args': { 'scenario': 'uas-480.xml', '-p': '5062'}}<br>- - { 'key-args': { 'scenario': 'uac-no-hangup.xml', '-d': '2000', '-s': 'unavailable', '-p': '5061'} }<br>-<br>-ami-config:<br>- -<br>- type: 'headermatch'<br>- conditions:<br>- match:<br>- Event: 'UserEvent'<br>- UserEvent: 'RedirectReason'<br>- requirements:<br>- match:<br>- Result: 'Success'<br>- count: '3'<br>-<br>-properties:<br>- minversion: '13.0.0'<br>- maxversion: '13.8.0'<br>- dependencies:<br>- - python: 'twisted'<br>- - python: 'starpy'<br>- - sipp:<br>- version: 'v3.1'<br>- - asterisk: 'app_dial'<br>- - asterisk: 'app_userevent'<br>- - asterisk: 'chan_sip'<br>- - asterisk: 'func_callerid'<br>- tags:<br>- - SIP<br>diff --git a/tests/channels/pjsip/refer_send_to_vm/v12.2.0/test-config.yaml b/tests/channels/pjsip/refer_send_to_vm/v12.2.0/test-config.yaml<br>deleted file mode 100644<br>index 0344df6..0000000<br>--- a/tests/channels/pjsip/refer_send_to_vm/v12.2.0/test-config.yaml<br>+++ /dev/null<br>@@ -1,54 +0,0 @@<br>-testinfo:<br>- summary: 'Test the send to voicemail headers in a refer.'<br>- description: |<br>- 'When using a Digium phone depending on the configuration it is<br>- possible for a REFER to contain a diversion and/or custom header.<br>- This tests that the appropriate variables are set on the channel<br>- before entering the dialplan when those headers are present in<br>- a REFER.'<br>-<br>-properties:<br>- minversion: '12.2.0'<br>- maxversion: '13.8.0'<br>- dependencies:<br>- - python: 'twisted'<br>- - python: 'starpy'<br>- - app: 'sipp'<br>- - asterisk: 'app_dial'<br>- - asterisk: 'app_userevent'<br>- - asterisk: 'func_callerid'<br>- - asterisk: 'res_pjsip'<br>- - asterisk: 'res_pjsip_header_funcs'<br>- - asterisk: 'res_pjsip_refer'<br>- - asterisk: 'res_pjsip_send_to_voicemail'<br>- tags:<br>- - pjsip<br>-<br>-test-modules:<br>- test-object:<br>- config-section: test-object-config<br>- typename: 'sipp.SIPpTestCase'<br>- modules:<br>- -<br>- config-section: ami-config<br>- typename: 'ami.AMIEventModule'<br>-<br>-test-object-config:<br>- test-iterations:<br>- -<br>- scenarios:<br>- - { 'key-args': { 'scenario':'refer.xml', '-p':'5062' } }<br>- - { 'key-args': { 'scenario':'invite.xml', '-p':'5061' } }<br>-<br>-ami-config:<br>- -<br>- id: '0'<br>- type: 'headermatch'<br>- count: '1'<br>- conditions:<br>- match:<br>- Event: 'UserEvent'<br>- requirements:<br>- match:<br>- Status: 'passed'<br>-<br></pre><p>To view, visit <a href="https://gerrit.asterisk.org/8045">change 8045</a>. 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<div style="display:none"> Gerrit-Project: testsuite </div>
<div style="display:none"> Gerrit-Branch: new_master </div>
<div style="display:none"> Gerrit-MessageType: newchange </div>
<div style="display:none"> Gerrit-Change-Id: I93793406961fed649f89a46c775932004be07769 </div>
<div style="display:none"> Gerrit-Change-Number: 8045 </div>
<div style="display:none"> Gerrit-PatchSet: 1 </div>
<div style="display:none"> Gerrit-Owner: Kevin Harwell <kharwell@digium.com> </div>