[Asterisk-code-review] channels/pjsip/ami/pjsip notify/channel: 3PCC patch for AMI ... (testsuite[13])
Richard Mudgett
asteriskteam at digium.com
Wed Apr 11 18:19:05 CDT 2018
Richard Mudgett has uploaded this change for review. ( https://gerrit.asterisk.org/8771
Change subject: channels/pjsip/ami/pjsip_notify/channel: 3PCC patch for AMI "PJSIPNotify"
......................................................................
channels/pjsip/ami/pjsip_notify/channel: 3PCC patch for AMI "PJSIPNotify"
ASTERISK-27697
Change-Id: I2047ca031c3c00b17b2ea06f1008fb17f84846d0
---
A tests/channels/pjsip/ami/pjsip_notify/channel/configs/ast1/extensions.conf
A tests/channels/pjsip/ami/pjsip_notify/channel/configs/ast1/pjsip.conf
A tests/channels/pjsip/ami/pjsip_notify/channel/sipp/callee.xml
A tests/channels/pjsip/ami/pjsip_notify/channel/sipp/caller.xml
A tests/channels/pjsip/ami/pjsip_notify/channel/test-config.yaml
M tests/channels/pjsip/ami/pjsip_notify/tests.yaml
6 files changed, 278 insertions(+), 0 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/71/8771/1
diff --git a/tests/channels/pjsip/ami/pjsip_notify/channel/configs/ast1/extensions.conf b/tests/channels/pjsip/ami/pjsip_notify/channel/configs/ast1/extensions.conf
new file mode 100644
index 0000000..35471e5
--- /dev/null
+++ b/tests/channels/pjsip/ami/pjsip_notify/channel/configs/ast1/extensions.conf
@@ -0,0 +1,3 @@
+[default]
+exten => callee,1,Dial(PJSIP/callee)
+
diff --git a/tests/channels/pjsip/ami/pjsip_notify/channel/configs/ast1/pjsip.conf b/tests/channels/pjsip/ami/pjsip_notify/channel/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..3215e02
--- /dev/null
+++ b/tests/channels/pjsip/ami/pjsip_notify/channel/configs/ast1/pjsip.conf
@@ -0,0 +1,39 @@
+[system]
+type=system
+timer_t1=100
+timer_b=6400
+
+[local-transport-udp]
+type=transport
+bind=127.0.0.1
+protocol=udp
+
+[caller]
+type=endpoint
+aors=caller
+context=default
+allow=!all,ulaw
+rewrite_contact=yes
+direct_media=no
+
+[caller]
+type=aor
+max_contacts=1
+minimum_expiration=5
+default_expiration=30
+contact=sip:caller at 127.0.0.1:5062
+
+[callee]
+type=endpoint
+aors=callee
+context=default
+allow=!all,ulaw
+rewrite_contact=yes
+direct_media=no
+
+[callee]
+type=aor
+max_contacts=1
+minimum_expiration=5
+default_expiration=30
+contact=sip:callee at 127.0.0.1:5063
diff --git a/tests/channels/pjsip/ami/pjsip_notify/channel/sipp/callee.xml b/tests/channels/pjsip/ami/pjsip_notify/channel/sipp/callee.xml
new file mode 100644
index 0000000..eee3f03
--- /dev/null
+++ b/tests/channels/pjsip/ami/pjsip_notify/channel/sipp/callee.xml
@@ -0,0 +1,115 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Notify Request with Call-ID">
+
+ <recv request="INVITE">
+ <action>
+ <ereg regexp=": .*"
+ search_in="hdr"
+ header="Call-ID"
+ check_it="true"
+ assign_to="1"/>
+ <ereg regexp=": .*"
+ search_in="hdr"
+ header="CSeq"
+ check_it="true"
+ assign_to="2"/>
+ <log message="Received INVITE with Call-ID [$1] and CSeq [$2]." />
+ </action>
+ </recv>
+
+ <nop>
+ <action>
+ <assignstr assign_to="3" value="[last_Via:]" />
+ </action>
+ </nop>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag[call_number]
+ Call-ID: [call_id]
+ [last_CSeq:]
+ Contact: <sip:user1@[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag[call_number]
+ Call-ID: [call_id]
+ [last_CSeq:]
+ Contact: <sip:user1@[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv request="NOTIFY">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ Call-ID: [call_id]
+ [last_CSeq:]
+ Contact: <sip:user1@[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [$3]
+ [last_From:]
+ [last_To:]
+ Call-ID: [call_id]
+ CSeq[$2]
+ Contact: <sip:user1@[local_ip]:[local_port];transport=[transport]>
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=user2 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[local_ip_type] [local_ip]
+ t=0 0
+ m=audio 9000 RTP/AVP 0
+ a=rtpmap:8 PCMU/8000
+ ]]>
+ </send>
+
+ <recv request="ACK">
+ </recv>
+
+ <recv request="BYE">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ Call-ID: [call_id]
+ [last_CSeq:]
+ Contact: <sip:user1@[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+ </send>
+</scenario>
diff --git a/tests/channels/pjsip/ami/pjsip_notify/channel/sipp/caller.xml b/tests/channels/pjsip/ami/pjsip_notify/channel/sipp/caller.xml
new file mode 100644
index 0000000..53d0781
--- /dev/null
+++ b/tests/channels/pjsip/ami/pjsip_notify/channel/sipp/caller.xml
@@ -0,0 +1,72 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Notify Request with Call-ID">
+
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:callee at voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: caller <sip:caller at voxbone.com>;tag=[call_number]
+ To: callee <sip:callee at voxbone.com:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[local_ip_type] [local_ip]
+ t=0 0
+ m=audio 9000 RTP/AVP 0
+ a=rtpmap:8 PCMU/8000
+ ]]>
+ </send>
+
+ <recv response="100" optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="200">
+ <action>
+ <ereg regexp=";tag=.*"
+ search_in="hdr"
+ header="To:"
+ check_it="true"
+ assign_to="1"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ ACK sip:callee at voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: caller <sip:caller at voxbone.com>;tag=[call_number]
+ To: callee <sip:callee at voxbone.com:[remote_port]>[$1]
+ Call-ID: [call_id]
+ CSeq: 2 ACK
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send>
+ <![CDATA[
+ BYE sip:callee at voxbone.com SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: caller <sip:caller at voxbone.com>;tag=[call_number]
+ To: callee <sip:callee at voxbone.com:[remote_port]>[$1]
+ Call-ID: [call_id]
+ CSeq: 3 BYE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv response="200">
+ </recv>
+</scenario>
diff --git a/tests/channels/pjsip/ami/pjsip_notify/channel/test-config.yaml b/tests/channels/pjsip/ami/pjsip_notify/channel/test-config.yaml
new file mode 100644
index 0000000..5d2bb64
--- /dev/null
+++ b/tests/channels/pjsip/ami/pjsip_notify/channel/test-config.yaml
@@ -0,0 +1,48 @@
+info:
+ summary: 'Test PJSIPNotify AMI Action for Channel'
+ description: |
+ This Tests the AMI Action PJSIPNotify with the
+ channel parameter given, generating an in-DIALOG
+ request.
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: sipp.SIPpTestCase
+ modules:
+ -
+ config-section: ami-config
+ typename: 'pluggable_modules.EventActionModule'
+
+test-object-config:
+ fail-on-any: True
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': { 'scenario': 'callee.xml', '-i': '127.0.0.1', '-p': '5063', '-trace_msg': '-pause_msg_ign' } }
+ - { 'key-args': { 'scenario': 'caller.xml', '-i': '127.0.0.1', '-p': '5062', '-trace_msg': '-pause_msg_ign' } }
+
+ami-config:
+ -
+ ami-events:
+ type: 'headermatch'
+ conditions:
+ match:
+ Event: 'Newstate'
+ Channel: 'PJSIP/callee-.*'
+ ChannelStateDesc: 'Ringing'
+ ami-actions:
+ action:
+ Action: 'PJSIPNotify'
+ Channel: '{channel}'
+ Variable: 'Event=talk'
+
+properties:
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ - asterisk : 'chan_pjsip'
+ - asterisk : 'app_dial'
+ tags:
+ - PJSIP
+
diff --git a/tests/channels/pjsip/ami/pjsip_notify/tests.yaml b/tests/channels/pjsip/ami/pjsip_notify/tests.yaml
index acfccde..2bb7308 100644
--- a/tests/channels/pjsip/ami/pjsip_notify/tests.yaml
+++ b/tests/channels/pjsip/ami/pjsip_notify/tests.yaml
@@ -4,3 +4,4 @@
- test: 'reserved_headers'
- test: 'content'
- test: 'to_uri'
+ - test: 'channel'
--
To view, visit https://gerrit.asterisk.org/8771
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-Project: testsuite
Gerrit-Branch: 13
Gerrit-MessageType: newchange
Gerrit-Change-Id: I2047ca031c3c00b17b2ea06f1008fb17f84846d0
Gerrit-Change-Number: 8771
Gerrit-PatchSet: 1
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: lvl <digium at lvlconsultancy.nl>
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