[Asterisk-code-review] confbridge talking: Add test that checks talking status with... (testsuite[15])
Jenkins2
asteriskteam at digium.com
Wed Mar 28 17:02:32 CDT 2018
Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/8619 )
Change subject: confbridge_talking: Add test that checks talking status with hold
......................................................................
confbridge_talking: Add test that checks talking status with hold
If a channel in a softmix bridge is put on hold then it's talking status is
cleared and an AMI event is raised. This test checks for that event and whether
or not the talking status has been set to "on" or "off" when the channel is
put on hold.
ASTERISK-27755
Change-Id: I99e343439c5a43f178fc466aa2978f6dfd1ab73a
---
A tests/apps/confbridge/confbridge_talking/configs/ast1/confbridge.conf
A tests/apps/confbridge/confbridge_talking/configs/ast1/extensions.conf
A tests/apps/confbridge/confbridge_talking/configs/ast1/pjsip.conf
A tests/apps/confbridge/confbridge_talking/sipp/invite.xml
A tests/apps/confbridge/confbridge_talking/test-config.yaml
M tests/apps/confbridge/tests.yaml
6 files changed, 261 insertions(+), 0 deletions(-)
Approvals:
Joshua Colp: Looks good to me, but someone else must approve
Benjamin Keith Ford: Looks good to me, but someone else must approve
Sean Bright: Looks good to me, approved
Jenkins2: Approved for Submit
diff --git a/tests/apps/confbridge/confbridge_talking/configs/ast1/confbridge.conf b/tests/apps/confbridge/confbridge_talking/configs/ast1/confbridge.conf
new file mode 100644
index 0000000..df304ab
--- /dev/null
+++ b/tests/apps/confbridge/confbridge_talking/configs/ast1/confbridge.conf
@@ -0,0 +1,13 @@
+[general]
+
+[default_bridge]
+type=bridge
+
+[default_user]
+type=user
+
+[alice]
+type=user
+quiet=yes
+announce_only_user=no
+talk_detection_events=yes
diff --git a/tests/apps/confbridge/confbridge_talking/configs/ast1/extensions.conf b/tests/apps/confbridge/confbridge_talking/configs/ast1/extensions.conf
new file mode 100644
index 0000000..9112e08
--- /dev/null
+++ b/tests/apps/confbridge/confbridge_talking/configs/ast1/extensions.conf
@@ -0,0 +1,5 @@
+[default]
+
+exten => conference,1,Answer()
+ same => n,ConfBridge(confbridge,,${CALLERID(num)})
+ same => n,Hangup()
diff --git a/tests/apps/confbridge/confbridge_talking/configs/ast1/pjsip.conf b/tests/apps/confbridge/confbridge_talking/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..27796f1
--- /dev/null
+++ b/tests/apps/confbridge/confbridge_talking/configs/ast1/pjsip.conf
@@ -0,0 +1,9 @@
+[transport]
+type=transport
+bind=127.0.0.1
+protocol=udp
+
+[alice]
+type=endpoint
+allow=!all,ulaw
+media_address=127.0.0.1
diff --git a/tests/apps/confbridge/confbridge_talking/sipp/invite.xml b/tests/apps/confbridge/confbridge_talking/sipp/invite.xml
new file mode 100644
index 0000000..14adeea
--- /dev/null
+++ b/tests/apps/confbridge/confbridge_talking/sipp/invite.xml
@@ -0,0 +1,170 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Invite and hold">
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:[service]@[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: "[service]" <sip:[service]@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: [cseq] INVITE
+ Contact: sip:alice@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Supported: replaces, 100rel, timer, norefersub
+ Session-Expires: 1800
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 12345 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:[service]@[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:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [call_id]
+ Max-Forwards: 70
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <pause milliseconds="3000"/>
+
+ <!-- Start playing audio to be "detected" -->
+ <nop>
+ <action>
+ <exec rtp_stream="lib/python/asterisk/audio.ulaw,3,0"/>
+ </action>
+ </nop>
+
+ <pause milliseconds="3000"/>
+
+ <!-- Now send a hold reinvite -->
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:[service]@[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: "[service]" <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ [last_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 IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 12345 RTP/AVP 0 101
+ a=sendonly
+ 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:[service]@[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:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [call_id]
+ Max-Forwards: 70
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <pause milliseconds="3000"/>
+
+ <!-- Then take off hold -->
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:[service]@[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: "[service]" <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ [last_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 IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 12345 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:[service]@[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:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [call_id]
+ Max-Forwards: 70
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <pause milliseconds="1000"/>
+
+ <send>
+ <![CDATA[
+ BYE sip:[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+ To: <sip:[service]@[remote_ip]>[peer_tag_param]
+ CSeq: [cseq] BYE
+ Call-ID: [call_id]
+ Contact: <sip:[service]@[local_ip]:[local_port]>
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+</scenario>
diff --git a/tests/apps/confbridge/confbridge_talking/test-config.yaml b/tests/apps/confbridge/confbridge_talking/test-config.yaml
new file mode 100644
index 0000000..3bd5612
--- /dev/null
+++ b/tests/apps/confbridge/confbridge_talking/test-config.yaml
@@ -0,0 +1,63 @@
+testinfo:
+ summary: 'Test talk detection in a confbridge'
+ description: |
+ This test makes sure that the appropriate 'ConfbridgeTalking' events are
+ received when talking is detected and when the phone is put on hold.
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+ modules:
+ -
+ config-section: 'ami-config'
+ typename: 'pluggable_modules.EventActionModule'
+
+test-object-config:
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'invite.xml', '-i': '127.0.0.1', '-p': '5061', '-s': 'conference'} }
+
+ami-config:
+ -
+ ami-events:
+ conditions:
+ match:
+ Event: 'ConfbridgeTalking'
+ Channel: 'PJSIP/alice-*'
+ TalkingStatus: 'on'
+ count: 2
+ -
+ ami-events:
+ conditions:
+ match:
+ Event: 'Hold'
+ Channel: 'PJSIP/alice-*'
+ count: 1
+ -
+ ami-events:
+ conditions:
+ match:
+ Event: 'ConfbridgeTalking'
+ Channel: 'PJSIP/alice-*'
+ TalkingStatus: 'off'
+ count: 1
+ -
+ ami-events:
+ conditions:
+ match:
+ Event: 'Unhold'
+ Channel: 'PJSIP/alice-*'
+ count: 1
+
+properties:
+ tags:
+ - confbridge
+ - apps
+ dependencies:
+ - python : 'twisted'
+ - python : 'starpy'
+ - asterisk : 'app_confbridge'
+ - asterisk : 'chan_pjsip'
+
diff --git a/tests/apps/confbridge/tests.yaml b/tests/apps/confbridge/tests.yaml
index 48e7fca..3008383 100644
--- a/tests/apps/confbridge/tests.yaml
+++ b/tests/apps/confbridge/tests.yaml
@@ -8,6 +8,7 @@
- test: 'confbridge_recording'
- test: 'confbridge_result'
- test: 'confbridge_start_muted'
+ - test: 'confbridge_talking'
- test: 'confbridge_timeout'
- test: 'confbridge_triple_lindy'
- test: 'confbridge_waitmarked_kick'
--
To view, visit https://gerrit.asterisk.org/8619
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-Project: testsuite
Gerrit-Branch: 15
Gerrit-MessageType: merged
Gerrit-Change-Id: I99e343439c5a43f178fc466aa2978f6dfd1ab73a
Gerrit-Change-Number: 8619
Gerrit-PatchSet: 1
Gerrit-Owner: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Benjamin Keith Ford <bford at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Sean Bright <sean.bright at gmail.com>
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