[Asterisk-code-review] Initial commit for new master testsuite branch (testsuite[new master])
Kevin Harwell
asteriskteam at digium.com
Wed Jan 24 16:23:51 CST 2018
Kevin Harwell has uploaded this change for review. ( https://gerrit.asterisk.org/8045
Change subject: Initial commit for new master testsuite branch
......................................................................
Initial commit for new master testsuite branch
This branch will become the new master testsuite branch once the branches are
deemed ready (the current master branch will be suitably renamed).
Since the the Asterisk master branch is what will be come Asterisk 16.0.0 this
patch removes all tests that are not supported by the current Asterisk master
branch. Subsequently, this patch removes all tests that have a minversion
greater than 16.0.0 or a maxversion less than 16.0.0.
Change-Id: I93793406961fed649f89a46c775932004be07769
---
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/45/8045/1
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/8045
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-Project: testsuite
Gerrit-Branch: new_master
Gerrit-MessageType: newchange
Gerrit-Change-Id: I93793406961fed649f89a46c775932004be07769
Gerrit-Change-Number: 8045
Gerrit-PatchSet: 1
Gerrit-Owner: Kevin Harwell <kharwell at digium.com>
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