[Asterisk-code-review] channels/pjsip/dialplan functions/pjsip channel2: Add reques... (testsuite[master])
Richard Mudgett
asteriskteam at digium.com
Mon Dec 11 15:21:23 CST 2017
Richard Mudgett has uploaded this change for review. ( https://gerrit.asterisk.org/7514
Change subject: channels/pjsip/dialplan_functions/pjsip_channel2: Add request-uri/dnid test.
......................................................................
channels/pjsip/dialplan_functions/pjsip_channel2: Add request-uri/dnid test.
Test PJSIP channel for CALLERID(dnid) and CHANNEL(pjsip,request_uri)
ASTERISK-27478
Change-Id: Ic3b98ceb212f5310a616d6bf373eac32adb1a395
---
A tests/channels/pjsip/dialplan_functions/pjsip_channel2/configs/ast1/extensions.conf
A tests/channels/pjsip/dialplan_functions/pjsip_channel2/configs/ast1/pjsip.conf
A tests/channels/pjsip/dialplan_functions/pjsip_channel2/sipp/incoming.xml
A tests/channels/pjsip/dialplan_functions/pjsip_channel2/test-config.yaml
M tests/channels/pjsip/dialplan_functions/tests.yaml
5 files changed, 186 insertions(+), 6 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/14/7514/1
diff --git a/tests/channels/pjsip/dialplan_functions/pjsip_channel2/configs/ast1/extensions.conf b/tests/channels/pjsip/dialplan_functions/pjsip_channel2/configs/ast1/extensions.conf
new file mode 100644
index 0000000..9a96461
--- /dev/null
+++ b/tests/channels/pjsip/dialplan_functions/pjsip_channel2/configs/ast1/extensions.conf
@@ -0,0 +1,28 @@
+[globals]
+ruri_user = 1234
+ruri_full = sip:${ruri_user}@127.0.0.1:5060\;transport=UDP\;lr
+
+[default]
+exten = _XXXX,1,NoOp()
+same = n,NoOp(EXTEN is "${EXTEN}")
+same = n,NoOp(CALLERID(dnid) is "${CALLERID(dnid)}")
+same = n,NoOp(CHANNEL(pjsip,request_uri) is "${CHANNEL(pjsip,request_uri)}")
+
+same = n,Answer()
+
+same = n(exten),NoOp()
+same = n,GotoIf($["${ruri_user}"="${EXTEN}"]?dnid)
+same = n,UserEvent(Failure,Result:EXTEN is "${EXTEN}" expected "${ruri_user}")
+
+same = n(dnid),NoOp()
+same = n,GotoIf($["${ruri_user}"="${CALLERID(dnid)}"]?ruri)
+same = n,UserEvent(Failure,Result:CALLERID(dnid) is "${CALLERID(dnid)}" expected "${ruri_user}")
+
+same = n(ruri),NoOp()
+same = n,GotoIf($["${ruri_full}"="${CHANNEL(pjsip,request_uri)}"]?done)
+same = n,UserEvent(Failure,Result:CHANNEL(pjsip,request_uri) is "${CHANNEL(pjsip,request_uri)}" expected "${ruri_full}")
+
+same = n(done),NoOp()
+same = n,UserEvent(Done)
+same = n,Hangup()
+
diff --git a/tests/channels/pjsip/dialplan_functions/pjsip_channel2/configs/ast1/pjsip.conf b/tests/channels/pjsip/dialplan_functions/pjsip_channel2/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..30b28ed
--- /dev/null
+++ b/tests/channels/pjsip/dialplan_functions/pjsip_channel2/configs/ast1/pjsip.conf
@@ -0,0 +1,9 @@
+[transport-udp]
+type = transport
+protocol = udp
+bind = 0.0.0.0
+
+[alice]
+type = endpoint
+context = default
+allow = !all,ulaw
diff --git a/tests/channels/pjsip/dialplan_functions/pjsip_channel2/sipp/incoming.xml b/tests/channels/pjsip/dialplan_functions/pjsip_channel2/sipp/incoming.xml
new file mode 100644
index 0000000..671741f
--- /dev/null
+++ b/tests/channels/pjsip/dialplan_functions/pjsip_channel2/sipp/incoming.xml
@@ -0,0 +1,80 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE with different Request-URI and To URI's">
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:1234@[remote_ip]:[remote_port];transport=[transport];lr SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [service] <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:4321@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+ ]]>
+ </send>
+
+ <recv response="100" optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+ ACK sip:1234@[remote_ip]:[remote_port];transport=[transport];lr SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [service] <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:4321@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv request="BYE">
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- definition of the response time repartition table (unit is ms) -->
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <!-- definition of the call length repartition table (unit is ms) -->
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
diff --git a/tests/channels/pjsip/dialplan_functions/pjsip_channel2/test-config.yaml b/tests/channels/pjsip/dialplan_functions/pjsip_channel2/test-config.yaml
new file mode 100644
index 0000000..bc1a537
--- /dev/null
+++ b/tests/channels/pjsip/dialplan_functions/pjsip_channel2/test-config.yaml
@@ -0,0 +1,62 @@
+testinfo:
+ summary: 'Test PJSIP channel for CALLERID(dnid) and CHANNEL(pjsip,request_uri)'
+ description: |
+ 'Run a SIPp scenario that places a call from endpoint alice to check
+ that the CALLERID(dnid) and CHANNEL(pjsip,request_uri) are set to
+ expected values.'
+
+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': 'incoming.xml', '-i': '127.0.0.1', '-p': '5061', '-s': 'alice'} }
+
+ami-config:
+ -
+ type: 'headermatch'
+ id: '0'
+ conditions:
+ match:
+ Event: 'Newchannel'
+ Channel: 'PJSIP/alice-.*'
+ count: '1'
+ -
+ type: 'headermatch'
+ id: '0'
+ conditions:
+ match:
+ Event: 'UserEvent'
+ Channel: 'PJSIP/alice-.*'
+ UserEvent: 'Failure'
+ count: '0'
+ -
+ type: 'headermatch'
+ id: '0'
+ conditions:
+ match:
+ Event: 'UserEvent'
+ Channel: 'PJSIP/alice-.*'
+ UserEvent: 'Done'
+ count: '1'
+
+properties:
+ minversion: ['13.19.0', '15.2.0']
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ - asterisk : 'app_dial'
+ - asterisk : 'app_userevent'
+ - asterisk : 'func_callerid'
+ - asterisk : 'func_channel'
+ - asterisk : 'res_pjsip'
+ tags:
+ - pjsip
diff --git a/tests/channels/pjsip/dialplan_functions/tests.yaml b/tests/channels/pjsip/dialplan_functions/tests.yaml
index f653aa8..a2e26a0 100644
--- a/tests/channels/pjsip/dialplan_functions/tests.yaml
+++ b/tests/channels/pjsip/dialplan_functions/tests.yaml
@@ -1,10 +1,11 @@
# Enter tests here in the order they should be considered for execution:
tests:
- - test: 'pjsip_channel'
- - test: 'pjsip_endpoint'
- - test: 'pjsip_aor'
- - test: 'pjsip_contact'
- - test: 'pjsip_session_refresh'
- test: 'chan_is_avail'
- - test: 'pjsip_header'
+ - test: 'pjsip_aor'
+ - test: 'pjsip_channel'
+ - test: 'pjsip_channel2'
+ - test: 'pjsip_contact'
- test: 'pjsip_dtmfmode'
+ - test: 'pjsip_endpoint'
+ - test: 'pjsip_header'
+ - test: 'pjsip_session_refresh'
--
To view, visit https://gerrit.asterisk.org/7514
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-MessageType: newchange
Gerrit-Change-Id: Ic3b98ceb212f5310a616d6bf373eac32adb1a395
Gerrit-Change-Number: 7514
Gerrit-PatchSet: 1
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
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