[asterisk-commits] Testsuite: channels/pjsip/basic calls/outgoing/off-nominal/c... (testsuite[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jul 1 10:37:53 CDT 2016
Anonymous Coward #1000019 has submitted this change and it was merged.
Change subject: Testsuite: channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled
......................................................................
Testsuite: channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled
Test scenerios created as a response to ASTERISK-25772
Change-Id: Ifa21978d292cef621bd7690ca891f4a869c506cd
---
A tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/configs/ast1/extensions.conf
A tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/configs/ast1/pjsip.conf
A tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_cancel_invalid_sdp.xml
A tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_cancel_missing_sdp.xml
A tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_cancel_valid_sdp.xml
A tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_invalid_sdp.xml
A tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_missing_sdp.xml
A tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/test-config.yaml
M tests/channels/pjsip/basic_calls/outgoing/off-nominal/tests.yaml
9 files changed, 836 insertions(+), 0 deletions(-)
Approvals:
Mark Michelson: Looks good to me, but someone else must approve
Anonymous Coward #1000019: Verified
Joshua Colp: Looks good to me, approved
diff --git a/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/configs/ast1/extensions.conf b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/configs/ast1/extensions.conf
new file mode 100644
index 0000000..a19a2ac
--- /dev/null
+++ b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/configs/ast1/extensions.conf
@@ -0,0 +1,34 @@
+[general]
+
+[default]
+
+exten = start,1,NoOp()
+same = n,Dial(PJSIP/buba,2)
+same = n,UserEvent(DialResult,status:buba-${DIALSTATUS})
+
+; Try to give time separation in the Asterisk logs between tests to avoid overlap.
+same = n,Wait(1)
+same = n,Dial(PJSIP/carl,2)
+same = n,UserEvent(DialResult,status:carl-${DIALSTATUS})
+
+; Try to give time separation in the Asterisk logs between tests to avoid overlap.
+same = n,Wait(1)
+same = n,Dial(PJSIP/dave,2,g)
+same = n,UserEvent(DialResult,status:dave-${DIALSTATUS})
+
+; Try to give time separation in the Asterisk logs between tests to avoid overlap.
+same = n,Wait(1)
+same = n,Dial(PJSIP/evan,10,g)
+same = n,UserEvent(DialResult,status:evan-${DIALSTATUS})
+
+; Try to give time separation in the Asterisk logs between tests to avoid overlap.
+same = n,Wait(1)
+same = n,Dial(PJSIP/fred,10,g)
+same = n,UserEvent(DialResult,status:fred-${DIALSTATUS})
+
+same = n,Hangup()
+
+exten = target,1,NoOp()
+same = n,Answer()
+same = n,Echo()
+same = n,Hangup()
diff --git a/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/configs/ast1/pjsip.conf b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..4ec7cfb
--- /dev/null
+++ b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/configs/ast1/pjsip.conf
@@ -0,0 +1,55 @@
+[transport]
+type=transport
+bind=127.0.0.1:5060
+protocol=udp
+
+[endpoint-template](!)
+type=endpoint
+context=default
+allow=!all,alaw,vp8
+direct_media=no
+identify_by=username
+
+[aor-template](!)
+type=aor
+
+
+[buba](endpoint-template)
+from_user=buba
+aors=buba
+
+[buba](aor-template)
+contact=sip:buba at 127.0.0.1:5071\;transport=udp
+
+
+[carl](endpoint-template)
+from_user=carl
+aors=carl
+
+[carl](aor-template)
+contact=sip:carl at 127.0.0.1:5072\;transport=udp
+
+
+[dave](endpoint-template)
+from_user=dave
+aors=dave
+
+[dave](aor-template)
+contact=sip:dave at 127.0.0.1:5073\;transport=udp
+
+
+[evan](endpoint-template)
+from_user=evan
+aors=evan
+
+[evan](aor-template)
+contact=sip:evan at 127.0.0.1:5074\;transport=udp
+
+
+[fred](endpoint-template)
+from_user=fred
+aors=fred
+
+[fred](aor-template)
+contact=sip:fred at 127.0.0.1:5075\;transport=udp
+
diff --git a/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_cancel_invalid_sdp.xml b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_cancel_invalid_sdp.xml
new file mode 100644
index 0000000..d0d2bcd
--- /dev/null
+++ b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_cancel_invalid_sdp.xml
@@ -0,0 +1,121 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="UAS for canceled call with invalid SDP answer test">
+ <Global variables="invite_from" />
+ <Global variables="invite_to" />
+ <Global variables="invite_callid" />
+ <Global variables="invite_cseq" />
+ <recv request="INVITE" crlf="true">
+ <action>
+ <!-- Save the from value. -->
+ <ereg regexp=".*(;tag=.*)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="invite_from"/>
+ <!-- Save the to value. -->
+ <ereg regexp=".*"
+ header="To:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="invite_to"/>
+ <!-- Save the callid value. -->
+ <ereg regexp=".*"
+ header="Call-ID:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="invite_callid"/>
+ <!-- Save the cseq value. -->
+ <ereg regexp=".*"
+ header="CSeq:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="invite_cseq"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv request="CANCEL" rtd="true" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ From:[$invite_from]
+ To:[$invite_to];tag=[pid]SIPpTag01[call_number]
+ Call-ID:[$invite_callid]
+ CSeq:[$invite_cseq]
+ Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=CGPLeg607266 362679584 181339793 IN IP[local_ip_type] [local_ip]
+ s=[sipp_version]
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 8 101
+ a=rtpmap:8 PCMA/8000
+ a=rtpmap:101 telephone-event/8000
+ a=fmtp:101 0-16
+ a=ptime:20
+ a=rtcpping:F:1253985:125398578
+ m=video [media_port+2] RTP/AVP 100
+ a=inactive
+ a=rtcpping:F:1253986:125398678
+ ]]>
+ </send>
+
+ <recv request="ACK" rtd="true" crlf="true">
+ </recv>
+
+ <recv request="BYE" rtd="true" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Linger awhile in case we get some unexpected message. -->
+ <pause/>
+ <pause/>
+
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
diff --git a/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_cancel_missing_sdp.xml b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_cancel_missing_sdp.xml
new file mode 100644
index 0000000..5498343
--- /dev/null
+++ b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_cancel_missing_sdp.xml
@@ -0,0 +1,105 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="UAS for canceled call with missing SDP answer test">
+ <Global variables="invite_from" />
+ <Global variables="invite_to" />
+ <Global variables="invite_callid" />
+ <Global variables="invite_cseq" />
+ <recv request="INVITE" crlf="true">
+ <action>
+ <!-- Save the from value. -->
+ <ereg regexp=".*(;tag=.*)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="invite_from"/>
+ <!-- Save the to value. -->
+ <ereg regexp=".*"
+ header="To:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="invite_to"/>
+ <!-- Save the callid value. -->
+ <ereg regexp=".*"
+ header="Call-ID:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="invite_callid"/>
+ <!-- Save the cseq value. -->
+ <ereg regexp=".*"
+ header="CSeq:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="invite_cseq"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv request="CANCEL" rtd="true" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ From:[$invite_from]
+ To:[$invite_to];tag=[pid]SIPpTag01[call_number]
+ Call-ID:[$invite_callid]
+ CSeq:[$invite_cseq]
+ Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv request="ACK" rtd="true" crlf="true">
+ </recv>
+
+ <recv request="BYE" rtd="true" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Linger awhile in case we get some unexpected message. -->
+ <pause/>
+ <pause/>
+
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
diff --git a/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_cancel_valid_sdp.xml b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_cancel_valid_sdp.xml
new file mode 100644
index 0000000..6a4cbaf
--- /dev/null
+++ b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_cancel_valid_sdp.xml
@@ -0,0 +1,119 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="UAS for canceled call with valid SDP answer test">
+ <Global variables="invite_from" />
+ <Global variables="invite_to" />
+ <Global variables="invite_callid" />
+ <Global variables="invite_cseq" />
+ <recv request="INVITE" crlf="true">
+ <action>
+ <!-- Save the from value. -->
+ <ereg regexp=".*(;tag=.*)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="invite_from"/>
+ <!-- Save the to value. -->
+ <ereg regexp=".*"
+ header="To:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="invite_to"/>
+ <!-- Save the callid value. -->
+ <ereg regexp=".*"
+ header="Call-ID:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="invite_callid"/>
+ <!-- Save the cseq value. -->
+ <ereg regexp=".*"
+ header="CSeq:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="invite_cseq"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv request="CANCEL" rtd="true" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ From:[$invite_from]
+ To:[$invite_to];tag=[pid]SIPpTag01[call_number]
+ Call-ID:[$invite_callid]
+ CSeq:[$invite_cseq]
+ Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=CGPLeg607266 362679584 181339793 IN IP[local_ip_type] [local_ip]
+ s=[sipp_version]
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 8 101
+ a=rtpmap:8 PCMA/8000
+ a=rtpmap:101 telephone-event/8000
+ a=fmtp:101 0-16
+ a=ptime:20
+ a=rtcpping:F:1253985:125398578
+ m=video 0 RTP/AVP 100
+ ]]>
+ </send>
+
+ <recv request="ACK" rtd="true" crlf="true">
+ </recv>
+
+ <recv request="BYE" rtd="true" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Linger awhile in case we get some unexpected message. -->
+ <pause/>
+ <pause/>
+
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
diff --git a/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_invalid_sdp.xml b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_invalid_sdp.xml
new file mode 100644
index 0000000..bdb8f58
--- /dev/null
+++ b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_invalid_sdp.xml
@@ -0,0 +1,122 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="UAS with invalid SDP answer test">
+ <Global variables="remote_tag" />
+ <recv request="INVITE" crlf="true">
+ <action>
+ <!-- Save the from tag. We'll need it when we send our BYE -->
+ <ereg regexp=".*(;tag=.*)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="remote_tag"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 183 Call progress
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
+ Supported: 100rel,timer,replaces,histinfo,precondition
+ Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=CGPLeg607266 362679584 181339793 IN IP[local_ip_type] [local_ip]
+ s=[sipp_version]
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 8 101
+ c=IN IP[media_ip_type] [media_ip]
+ a=rtpmap:8 PCMA/8000
+ a=rtpmap:101 telephone-event/8000
+ a=fmtp:101 0-16
+ a=ptime:20
+ a=rtcpping:F:1253985:125398578
+ m=video [media_port+2] RTP/AVP 100
+ c=IN IP[media_ip_type] [media_ip]
+ a=inactive
+ a=rtcpping:F:1253986:125398678
+ ]]>
+ </send>
+
+ <pause/>
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=CGPLeg607266 362679584 181339793 IN IP[local_ip_type] [local_ip]
+ s=[sipp_version]
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 8 101
+ c=IN IP[media_ip_type] [media_ip]
+ a=rtpmap:8 PCMA/8000
+ a=rtpmap:101 telephone-event/8000
+ a=fmtp:101 0-16
+ a=ptime:20
+ a=rtcpping:F:1253985:125398578
+ m=video [media_port+2] RTP/AVP 100
+ c=IN IP[media_ip_type] [media_ip]
+ a=inactive
+ a=rtcpping:F:1253986:125398678
+ ]]>
+ </send>
+
+ <recv request="ACK" rtd="true" crlf="true">
+ </recv>
+
+ <recv request="BYE" rtd="true" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Linger awhile in case we get some unexpected message. -->
+ <pause/>
+ <pause/>
+
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
diff --git a/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_missing_sdp.xml b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_missing_sdp.xml
new file mode 100644
index 0000000..74e65fd
--- /dev/null
+++ b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/sipp/call_missing_sdp.xml
@@ -0,0 +1,71 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="UAS with missing SDP answer test">
+ <Global variables="remote_tag" />
+ <recv request="INVITE" crlf="true">
+ <action>
+ <!-- Save the from tag. We'll need it when we send our BYE -->
+ <ereg regexp=".*(;tag=.*)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="remote_tag"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <pause/>
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv request="ACK" rtd="true" crlf="true">
+ </recv>
+
+ <recv request="BYE" rtd="true" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Linger awhile in case we get some unexpected message. -->
+ <pause/>
+ <pause/>
+
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
diff --git a/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/test-config.yaml b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/test-config.yaml
new file mode 100644
index 0000000..a2edcd9
--- /dev/null
+++ b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/call_canceled/test-config.yaml
@@ -0,0 +1,208 @@
+testinfo:
+ summary: 'Test outgoing calls that are canceled'
+ description: |
+ 'There are five scenarios being tested.
+ 1) Call Buba and cancel the call before Buba answers. Buba "answers"
+ right as we cancel the call with a valid SDP response. This is the
+ CANCEL race condition described in RFC5407 section 3.1.2. In this
+ case PJPROJECT does not send the BYE for us so Asterisk must detect
+ this case and send the needed BYE.
+ 2) Call Carl and cancel the call before Carl answers. Carl "answers"
+ right as we cancel the call with an invalid SDP response. This is
+ the CANCEL race condition described in RFC5407 section 3.1.2. In
+ this case PJPROJECT does send the BYE for us because of the invalid
+ SDP response so Asterisk must detect this case and not send a BYE.
+ 3) Call Dave and cancel the call before Dave answers. Dave "answers"
+ right as we cancel the call with a missing SDP response. This is
+ the CANCEL race condition described in RFC5407 section 3.1.2. In
+ this case PJPROJECT does send the BYE for us because of the missing
+ SDP response so Asterisk must detect this case and not send a BYE.
+ 4) Call Evan and wait for Evan to answer. When Evan answers he sends
+ an invalid SDP response. In this case PJPROJECT cancels the call
+ because of the invalid SDP response and sends the BYE. Asterisk
+ must detect this case and not send a BYE.'
+ 5) Call Fred and wait for Fred to answer. When Fred answers he sends
+ a missing SDP response. In this case PJPROJECT cancels the call
+ because of the missing SDP response and sends the BYE. Asterisk
+ must detect this case and not send a BYE.'
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+ modules:
+ - config-section: ami-config
+ typename: 'ami.AMIEventModule'
+ - config-section: start-call
+ typename: 'pluggable_modules.Originator'
+
+test-object-config:
+ fail-on-any: True
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'-i': '127.0.0.1', '-p': '5071', '-s': 'buba', '-d': '500', 'scenario': 'call_cancel_valid_sdp.xml'} }
+ - { 'key-args': {'-i': '127.0.0.1', '-p': '5072', '-s': 'carl', '-d': '500', 'scenario': 'call_cancel_invalid_sdp.xml'} }
+ - { 'key-args': {'-i': '127.0.0.1', '-p': '5073', '-s': 'dave', '-d': '500', 'scenario': 'call_cancel_missing_sdp.xml'} }
+ - { 'key-args': {'-i': '127.0.0.1', '-p': '5074', '-s': 'evan', '-d': '500', 'scenario': 'call_invalid_sdp.xml'} }
+ - { 'key-args': {'-i': '127.0.0.1', '-p': '5075', '-s': 'fred', '-d': '500', 'scenario': 'call_missing_sdp.xml'} }
+
+start-call:
+ trigger: 'scenario_start'
+ scenario-trigger-after: '3'
+ channel: 'Local/start at default'
+ context: 'default'
+ exten: 'target'
+ priority: 1
+ async: True
+
+ami-config:
+ -
+ type: 'headermatch'
+ id: '0'
+ conditions:
+ match:
+ Event: 'UserEvent'
+ UserEvent: 'DialResult'
+ count: '5'
+ # Buba events
+ -
+ type: 'headermatch'
+ id: '0'
+ conditions:
+ match:
+ Event: 'UserEvent'
+ UserEvent: 'DialResult'
+ status: 'buba-.*'
+ requirements:
+ match:
+ status: 'buba-NOANSWER'
+ count: '1'
+ -
+ type: 'headermatch'
+ id: '0'
+ conditions:
+ match:
+ Event: 'TestEvent'
+ State: 'PJSIP_SESSION_CANCELED'
+ Endpoint: 'buba'
+ requirements:
+ match:
+ SDP: 'complete'
+ count: '1'
+ # Carl events
+ -
+ type: 'headermatch'
+ id: '0'
+ conditions:
+ match:
+ Event: 'UserEvent'
+ UserEvent: 'DialResult'
+ status: 'carl-.*'
+ requirements:
+ match:
+ status: 'carl-NOANSWER'
+ count: '1'
+ -
+ type: 'headermatch'
+ id: '0'
+ conditions:
+ match:
+ Event: 'TestEvent'
+ State: 'PJSIP_SESSION_CANCELED'
+ Endpoint: 'carl'
+ requirements:
+ match:
+ SDP: 'incomplete'
+ count: '1'
+ # Dave events
+ -
+ type: 'headermatch'
+ id: '0'
+ conditions:
+ match:
+ Event: 'UserEvent'
+ UserEvent: 'DialResult'
+ status: 'dave-.*'
+ requirements:
+ match:
+ status: 'dave-NOANSWER'
+ count: '1'
+ -
+ type: 'headermatch'
+ id: '0'
+ conditions:
+ match:
+ Event: 'TestEvent'
+ State: 'PJSIP_SESSION_CANCELED'
+ Endpoint: 'dave'
+ requirements:
+ match:
+ SDP: 'incomplete'
+ count: '1'
+ # Evan events
+ -
+ type: 'headermatch'
+ id: '0'
+ conditions:
+ match:
+ Event: 'UserEvent'
+ UserEvent: 'DialResult'
+ status: 'evan-.*'
+ requirements:
+ match:
+ status: 'evan-ANSWER'
+ count: '1'
+ -
+ type: 'headermatch'
+ id: '0'
+ conditions:
+ match:
+ Event: 'TestEvent'
+ State: 'PJSIP_SESSION_CANCELED'
+ Endpoint: 'evan'
+ requirements:
+ match:
+ SDP: 'incomplete'
+ count: '1'
+ # Fred events
+ -
+ type: 'headermatch'
+ id: '0'
+ conditions:
+ match:
+ Event: 'UserEvent'
+ UserEvent: 'DialResult'
+ status: 'fred-.*'
+ requirements:
+ match:
+ status: 'fred-ANSWER'
+ count: '1'
+ -
+ type: 'headermatch'
+ id: '0'
+ conditions:
+ match:
+ Event: 'TestEvent'
+ State: 'PJSIP_SESSION_CANCELED'
+ Endpoint: 'fred'
+ requirements:
+ match:
+ SDP: 'incomplete'
+ count: '1'
+
+properties:
+ minversion: '13.11.0'
+ dependencies:
+ - sipp:
+ version: 'v3.0'
+ - python: 'twisted'
+ - python: 'starpy'
+ - asterisk: 'app_dial'
+ - asterisk: 'app_echo'
+ - asterisk: 'app_userevent'
+ - asterisk: 'chan_pjsip'
+ - asterisk: 'res_pjsip'
+ tags:
+ - pjsip
+
diff --git a/tests/channels/pjsip/basic_calls/outgoing/off-nominal/tests.yaml b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/tests.yaml
index 0d72645..45fcb12 100644
--- a/tests/channels/pjsip/basic_calls/outgoing/off-nominal/tests.yaml
+++ b/tests/channels/pjsip/basic_calls/outgoing/off-nominal/tests.yaml
@@ -3,3 +3,4 @@
- test: 'bob_does_not_answer'
- test: 'bob_is_busy'
- test: 'bob_incompatible_codecs'
+ - test: 'call_canceled'
--
To view, visit https://gerrit.asterisk.org/3122
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Gerrit-MessageType: merged
Gerrit-Change-Id: Ifa21978d292cef621bd7690ca891f4a869c506cd
Gerrit-PatchSet: 1
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
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