[Asterisk-code-review] Initial commit for 15 testsuite branch (testsuite[15])

[Kevin Harwell]

Kevin Harwell has uploaded this change for review. ( https://gerrit.asterisk.org/8044


Change subject: Initial commit for 15 testsuite branch
......................................................................

Initial commit for 15 testsuite branch

This patch removes all tests that are not supported by the current Asterisk 15
branch. Subsequently, this patch removes all tests that have a minversion
greater than 15.3.0 or a maxversion less than 15.3.0.

Change-Id: I82c11fd46d2722837969724d0779bfd81fc873b4
---
D tests/channels/SIP/redirecting_reason/v13.0.0/test-config.yaml
D tests/channels/pjsip/refer_send_to_vm/v12.2.0/test-config.yaml
2 files changed, 0 insertions(+), 121 deletions(-)



  git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/44/8044/1

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
deleted file mode 100644
index 15bb36a..0000000
--- a/tests/channels/SIP/redirecting_reason/v13.0.0/test-config.yaml
+++ /dev/null
@@ -1,67 +0,0 @@
-testinfo:
-    summary: 'Test that custom SIP redirecting reasons are accessible from the dialplan'
-    description: |
-        'This test performs three calls.
-
-         On the first call, an INVITE with a Diversion header arrives into Asterisk. We
-         ensure that from the dialplan, the redirecting reason is what we expect it to be.
-
-         On the second call, we place an outgoing call to a UAS that responds with a 302. In
-         that 302, there is a Diversion header with a reason parameter. We again check in the
-         dialplan to ensure that the redirecting reason has been set to this value.
-
-         On the third call, we place an outgoing call to a UAS that responds with a 480, and
-         with custom status text. We then ensure that the custom status text appears as the
-         redirecting reason in the dialplan.'
-
-test-modules:
-    test-object:
-        config-section: 'calls'
-        typename: 'sipp.SIPpTestCase'
-    modules:
-        -
-            config-section: 'ami-config'
-            typename: 'ami.AMIEventModule'
-
-calls:
-    fail-on-any: False
-    test-iterations:
-        -
-            scenarios:
-                - { 'key-args': { 'scenario': 'uac-diversion.xml', '-s': 'test_diversion'} }
-        -
-            scenarios:
-                - { 'key-args': { 'scenario': 'uas-redirect.xml', '-p': '5062'},
-                    'ordered-args': ['-key', 'redir_target', 'test_diversion']}
-                - { 'key-args': { 'scenario': 'uac-no-hangup.xml', '-d': '2000', '-s': 'diverter', '-p': '5061'} }
-        -
-            scenarios:
-                - { 'key-args': { 'scenario': 'uas-480.xml', '-p': '5062'}}
-                - { 'key-args': { 'scenario': 'uac-no-hangup.xml', '-d': '2000', '-s': 'unavailable', '-p': '5061'} }
-
-ami-config:
-    -
-        type: 'headermatch'
-        conditions:
-            match:
-                Event: 'UserEvent'
-                UserEvent: 'RedirectReason'
-        requirements:
-            match:
-                Result: 'Success'
-        count: '3'
-
-properties:
-    minversion: '13.0.0'
-    maxversion: '13.8.0'
-    dependencies:
-        - python: 'twisted'
-        - python: 'starpy'
-        - sipp:
-            version: 'v3.1'
-        - asterisk: 'app_dial'
-        - asterisk: 'app_userevent'
-        - asterisk: 'chan_sip'
-        - asterisk: 'func_callerid'
-    tags:
-        - SIP
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
deleted file mode 100644
index 0344df6..0000000
--- a/tests/channels/pjsip/refer_send_to_vm/v12.2.0/test-config.yaml
+++ /dev/null
@@ -1,54 +0,0 @@
-testinfo:
-    summary: 'Test the send to voicemail headers in a refer.'
-    description: |
-        'When using a Digium phone depending on the configuration it is
-         possible for a REFER to contain a diversion and/or custom header.
-         This tests that the appropriate variables are set on the channel
-         before entering the dialplan when those headers are present in
-         a REFER.'
-
-properties:
-    minversion: '12.2.0'
-    maxversion: '13.8.0'
-    dependencies:
-         - python: 'twisted'
-         - python: 'starpy'
-         - app: 'sipp'
-         - asterisk: 'app_dial'
-         - asterisk: 'app_userevent'
-         - asterisk: 'func_callerid'
-         - asterisk: 'res_pjsip'
-         - asterisk: 'res_pjsip_header_funcs'
-         - asterisk: 'res_pjsip_refer'
-         - asterisk: 'res_pjsip_send_to_voicemail'
-    tags:
-        - pjsip
-
-test-modules:
-    test-object:
-        config-section: test-object-config
-        typename: 'sipp.SIPpTestCase'
-    modules:
-        -
-            config-section: ami-config
-            typename: 'ami.AMIEventModule'
-
-test-object-config:
-    test-iterations:
-        -
-             scenarios:
-                - { 'key-args': { 'scenario':'refer.xml', '-p':'5062' } }
-                - { 'key-args': { 'scenario':'invite.xml', '-p':'5061' } }
-
-ami-config:
-    -
-        id: '0'
-        type: 'headermatch'
-        count: '1'
-        conditions:
-            match:
-                Event: 'UserEvent'
-        requirements:
-            match:
-                Status: 'passed'
-

-- 
To view, visit https://gerrit.asterisk.org/8044
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-Project: testsuite
Gerrit-Branch: 15
Gerrit-MessageType: newchange
Gerrit-Change-Id: I82c11fd46d2722837969724d0779bfd81fc873b4
Gerrit-Change-Number: 8044
Gerrit-PatchSet: 1
Gerrit-Owner: Kevin Harwell <kharwell at digium.com>
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