[Asterisk-code-review] app_senddtmf: Add receive mode to AMI Action PlayDTMF (testsuite[16])
lvl
asteriskteam at digium.com
Thu Nov 7 11:09:48 CST 2019
lvl has uploaded this change for review. ( https://gerrit.asterisk.org/c/testsuite/+/13164 )
Change subject: app_senddtmf: Add receive mode to AMI Action PlayDTMF
......................................................................
app_senddtmf: Add receive mode to AMI Action PlayDTMF
Change-Id: I50f29a7c86ad10ef0f94d1aebb6eca4d905b4ec4
---
A tests/manager/playdtmf/configs/ast1/extensions.conf
A tests/manager/playdtmf/configs/ast1/pjsip.conf
A tests/manager/playdtmf/sipp/invite.xml
A tests/manager/playdtmf/test-config.yaml
M tests/manager/tests.yaml
5 files changed, 175 insertions(+), 0 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/64/13164/1
diff --git a/tests/manager/playdtmf/configs/ast1/extensions.conf b/tests/manager/playdtmf/configs/ast1/extensions.conf
new file mode 100644
index 0000000..d673b0f
--- /dev/null
+++ b/tests/manager/playdtmf/configs/ast1/extensions.conf
@@ -0,0 +1,13 @@
+[default]
+exten => dialplan,1,Answer()
+ same => n,UserEvent(ready)
+ same => n,WaitExten(2)
+ same => n,Hangup()
+
+exten => 1,1,UserEvent(invalid)
+ same => n,Wait(1)
+ same => n,Hangup()
+
+exten => 2,1,UserEvent(valid)
+ same => n,Wait(1)
+ same => n,Hangup()
diff --git a/tests/manager/playdtmf/configs/ast1/pjsip.conf b/tests/manager/playdtmf/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..b5a8e95
--- /dev/null
+++ b/tests/manager/playdtmf/configs/ast1/pjsip.conf
@@ -0,0 +1,33 @@
+[system]
+type=system
+timer_t1=100
+timer_b=6400
+
+[local]
+type=transport
+protocol=udp
+bind=0.0.0.0
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[endpoint_t](!)
+type=endpoint
+context=default
+transport=local
+direct_media=no
+disallow=all
+allow=ulaw
+
+[aor_t](!)
+type=aor
+max_contacts=1
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+;;; alice
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[alice](aor_t)
+contact=sip:alice at 127.0.0.1:5061
+
+[alice](endpoint_t)
+aors=alice
diff --git a/tests/manager/playdtmf/sipp/invite.xml b/tests/manager/playdtmf/sipp/invite.xml
new file mode 100644
index 0000000..3aa13c7
--- /dev/null
+++ b/tests/manager/playdtmf/sipp/invite.xml
@@ -0,0 +1,71 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Invite with variables">
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:dialplan@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: "alice" <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+ To: "bob" <sip:dialplan@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: [cseq] INVITE
+ Contact: sip:alice@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP4 [local_ip]
+ s=-
+ c=IN IP4 [local_ip]
+ t=0 0
+ m=audio 2226 RTP/AVP 0 101
+ a=sendrecv
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true" />
+
+ <recv response="180" optional="true" />
+
+ <recv response="200" />
+
+ <send>
+ <![CDATA[
+ ACK sip:dialplan@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:alice@[local_ip]>;tag=[call_number]
+ To: <sip:dialplan@[remote_ip]:[remote_port]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [call_id]
+ Contact: <sip:alice@[local_ip]>
+ Allow: INVITE, ACK, MESSAGE, BYE
+ Max-Forwards: 70
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="BYE" crlf="true" />
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:alice@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, MESSAGE, BYE
+ Content-Type: application/sdp
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+</scenario>
diff --git a/tests/manager/playdtmf/test-config.yaml b/tests/manager/playdtmf/test-config.yaml
new file mode 100644
index 0000000..958587f
--- /dev/null
+++ b/tests/manager/playdtmf/test-config.yaml
@@ -0,0 +1,57 @@
+testinfo:
+ summary: 'PlayDTMF testcase'
+ description: |
+ 'asdf'
+
+properties:
+ dependencies:
+ - app : 'sipp'
+ - asterisk : 'res_pjsip'
+ tags:
+ - pjsip
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+ modules:
+ -
+ config-section: 'ami-config'
+ typename: 'pluggable_modules.EventActionModule'
+
+test-object-config:
+ memcheck-delay-stop: 7
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': { 'scenario':'invite.xml', '-p':'5061' } }
+
+ami-config:
+ -
+ ami-events:
+ count: 1
+ conditions:
+ match:
+ Event: 'UserEvent'
+ UserEvent: 'ready'
+ ami-actions:
+ # without the "Receive" parameter, DTMF will be sent out on this channel and won't impact the PBX/dialplan side
+ -
+ action:
+ Action: 'PlayDtmf'
+ Channel: '{channel}'
+ Digit: '1'
+ # with the "Receive" parameter, DTMF will be received on this channel and be caught by WaitExten() in the dialplan
+ -
+ action:
+ Action: 'PlayDtmf'
+ Channel: '{channel}'
+ Digit: '2'
+ Receive: '1'
+ -
+ ami-events:
+ count: 1
+ conditions:
+ match:
+ Event: 'UserEvent'
+ UserEvent: 'valid'
diff --git a/tests/manager/tests.yaml b/tests/manager/tests.yaml
index eab0ef4..6e6f8d5 100644
--- a/tests/manager/tests.yaml
+++ b/tests/manager/tests.yaml
@@ -23,3 +23,4 @@
- test: 'response-time'
- test: 'escaped_values'
- dir: 'redirect'
+ - test: 'playdtmf'
--
To view, visit https://gerrit.asterisk.org/c/testsuite/+/13164
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Gerrit-Project: testsuite
Gerrit-Branch: 16
Gerrit-Change-Id: I50f29a7c86ad10ef0f94d1aebb6eca4d905b4ec4
Gerrit-Change-Number: 13164
Gerrit-PatchSet: 1
Gerrit-Owner: lvl <digium at lvlconsultancy.nl>
Gerrit-MessageType: newchange
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