[asterisk-commits] testsuite: Test 180 & 183 with and without sdp (testsuite[19])

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Apr 26 18:49:12 CDT 2022


Friendly Automation has submitted this change. ( https://gerrit.asterisk.org/c/testsuite/+/18447 )

Change subject: testsuite: Test 180 & 183 with and without sdp
......................................................................

testsuite: Test 180 & 183 with and without sdp

Add tests to ensure we handle 180 after a 183

ASTERISK-29842

Change-Id: Ie0ba7dd79f881110fc7d3f000e72dfc52dd7edf3
---
A tests/channels/pjsip/allow_sending_180_after_183/disabled/configs/ast1/extensions.conf
A tests/channels/pjsip/allow_sending_180_after_183/disabled/configs/ast1/pjsip.conf
A tests/channels/pjsip/allow_sending_180_after_183/disabled/sipp/A_PARTY.xml
A tests/channels/pjsip/allow_sending_180_after_183/disabled/sipp/B_PARTY.xml
A tests/channels/pjsip/allow_sending_180_after_183/disabled/test-config.yaml
A tests/channels/pjsip/allow_sending_180_after_183/enabled/configs/ast1/extensions.conf
A tests/channels/pjsip/allow_sending_180_after_183/enabled/configs/ast1/pjsip.conf
A tests/channels/pjsip/allow_sending_180_after_183/enabled/sipp/A_PARTY.xml
A tests/channels/pjsip/allow_sending_180_after_183/enabled/sipp/B_PARTY.xml
A tests/channels/pjsip/allow_sending_180_after_183/enabled/test-config.yaml
A tests/channels/pjsip/allow_sending_180_after_183/tests.yaml
M tests/channels/pjsip/tests.yaml
12 files changed, 560 insertions(+), 0 deletions(-)

Approvals:
  Kevin Harwell: Looks good to me, approved
  Friendly Automation: Approved for Submit



diff --git a/tests/channels/pjsip/allow_sending_180_after_183/disabled/configs/ast1/extensions.conf b/tests/channels/pjsip/allow_sending_180_after_183/disabled/configs/ast1/extensions.conf
new file mode 100644
index 0000000..ad7b155
--- /dev/null
+++ b/tests/channels/pjsip/allow_sending_180_after_183/disabled/configs/ast1/extensions.conf
@@ -0,0 +1,11 @@
+[general]
+static=yes
+writeprotect=yes
+autofallthrough=yes
+clearglobalvars=no
+priorityjumping=yes
+
+[globals]
+
+[default]
+exten => _X.,1,Dial(pjsip/sbc,180)
diff --git a/tests/channels/pjsip/allow_sending_180_after_183/disabled/configs/ast1/pjsip.conf b/tests/channels/pjsip/allow_sending_180_after_183/disabled/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..caa4534
--- /dev/null
+++ b/tests/channels/pjsip/allow_sending_180_after_183/disabled/configs/ast1/pjsip.conf
@@ -0,0 +1,57 @@
+;--
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements start
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[general]
+sipdebug = yes
+
+[PEER_A]
+port = 5061
+
+[sbc]
+port = 5700
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements end
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+--;
+
+
+[global]
+type = global
+debug = yes
+allow_sending_180_after_183=no
+
+[transport-udp]
+type = transport
+protocol = udp
+bind = 0.0.0.0:5060
+
+[PEER_A]
+type = aor
+contact = sip:127.0.0.1:5061
+
+[PEER_A]
+type = identify
+endpoint = PEER_A
+match = 127.0.0.1
+
+[PEER_A]
+type = endpoint
+context = default
+allow = ulaw
+direct_media = no
+aors = PEER_A
+
+[sbc]
+type = aor
+contact = sip:127.0.0.1:5700
+
+[sbc]
+type = endpoint
+context = callcontrol
+allow = ulaw
+direct_media = no
+aors = sbc
+
diff --git a/tests/channels/pjsip/allow_sending_180_after_183/disabled/sipp/A_PARTY.xml b/tests/channels/pjsip/allow_sending_180_after_183/disabled/sipp/A_PARTY.xml
new file mode 100644
index 0000000..c2c3e7f
--- /dev/null
+++ b/tests/channels/pjsip/allow_sending_180_after_183/disabled/sipp/A_PARTY.xml
@@ -0,0 +1,96 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<!-- This program is free software; you can redistribute it and/or      -->
+<!-- modify it under the terms of the GNU General Public License as     -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version.                    -->
+<!--                                                                    -->
+<!-- This program is distributed in the hope that it will be useful,    -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
+<!-- GNU General Public License for more details.                       -->
+<!--                                                                    -->
+<!-- You should have received a copy of the GNU General Public License  -->
+<!-- along with this program; if not, write to the                      -->
+<!-- Free Software Foundation, Inc.,                                    -->
+<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
+<!--                                                                    -->
+<!--                 Sipp 'uac' scenario with pcap (rtp) play           -->
+<!--                                                                    -->
+<scenario name="CONTENT_TYPE_PARAMS">
+	<!-- In client mode (sipp placing calls), the Call-ID MUST be		-->
+	<!-- generated by sipp. To do so, use [call_id] keyword.		-->
+	<send retrans="500"><![CDATA[
+
+			INVITE sip:[service]@uni-tel.dk SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: sipp <sip:test at uni-tel.dk>;tag=[call_number]
+			To: mpe <sip:[service]@uni-tel.dk:[remote_port]>
+			Call-ID: [call_id]
+			CSeq: 1 INVITE
+			Contact: sip:sipp@[local_ip]:[local_port]
+			Max-Forwards: 70
+			Subject: Performance Test
+			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:0 PCMU/8000
+
+		]]></send>
+	<recv response="100" optional="true"/>
+	<recv response="183">
+		<action>
+			<ereg regexp="application/sdp" search_in="hdr" header="Content-Type:" check_it="true" assign_to="1"/>
+		</action>
+	</recv>
+	<Reference variables="1"/>
+	<recv response="183" optional="true">
+		<action>
+			<ereg regexp="application/sdp" search_in="hdr" header="Content-Type:" check_it="true" assign_to="2"/>
+		</action>
+	</recv>
+	<Reference variables="2"/>
+	<!-- By adding rrs="true" (Record Route Sets), the route sets		-->
+	<!-- are saved and used for following messages sent. Useful to test	-->
+	<!-- against stateful SIP proxies/B2BUAs.				-->
+	<recv response="200" rtd="true" crlf="true"/>
+	<!-- Packet lost can be simulated in any send/recv message by		-->
+	<!-- by adding the 'lost = "10"'. Value can be [1-100] percent.		-->
+	<send><![CDATA[
+
+			ACK sip:[service]@uni-tel.dk SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: sipp <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+			To: mpe <sip:[service]@uni-tel.dk>[peer_tag_param]
+			Call-ID: [call_id]
+			CSeq: 1 ACK
+			Contact: sip:sipp@[local_ip]:[local_port]
+			Max-Forwards: 70
+			Subject: Performance Test
+			Content-Length: 0
+
+		]]></send>
+	<pause milliseconds="1000"/>
+	<!-- The 'crlf' option inserts a blank line in the statistics report.	-->
+	<send retrans="500"><![CDATA[
+
+			BYE sip:[service]@uni-tel.dk SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: sipp <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+			To: mpe <sip:[service]@uni-tel.dk>[peer_tag_param]
+			Call-ID: [call_id]
+			CSeq: 2 BYE
+			Contact: sip:sipp@[local_ip]:[local_port]
+			Max-Forwards: 70
+			Subject: Performance Test
+			Content-Length: 0
+
+		]]></send>
+	<recv response="200" crlf="true"/>
+</scenario>
diff --git a/tests/channels/pjsip/allow_sending_180_after_183/disabled/sipp/B_PARTY.xml b/tests/channels/pjsip/allow_sending_180_after_183/disabled/sipp/B_PARTY.xml
new file mode 100644
index 0000000..ff8270b
--- /dev/null
+++ b/tests/channels/pjsip/allow_sending_180_after_183/disabled/sipp/B_PARTY.xml
@@ -0,0 +1,87 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<scenario name="CONTENT_TYPE_PARAMS">
+	<User variables="dummy"/>
+	<recv request="INVITE" crlf="true" rrs="true">
+		<action>
+			<ereg regexp=".*sip:asterisk at .*" search_in="hdr" header="Contact:" check_it="true" assign_to="dummy"/>
+		</action>
+	</recv>
+	<Reference variables="dummy"/>
+	<send><![CDATA[
+
+			SIP/2.0 100 Trying
+			[last_Via:]
+			[last_Call-ID:]
+			[last_From:]
+			[last_To:]
+			[last_CSeq:]
+			Content-Length: 0
+
+		]]></send>
+	<send><![CDATA[
+
+			SIP/2.0 183 Session Progress
+			[last_Via:]
+			[last_Call-ID:]
+			[last_From:]
+			[last_To:]
+			[last_CSeq:]
+			Content-Type: application/sdp
+			Content-Length: [len]
+
+			v=0
+			o=HuaweiSoftX3000 6644052 6644052 IN IP[local_ip_type] [local_ip]
+			s=Sip Call
+			c=IN IP[local_ip_type] [local_ip]
+			t=0 0
+			m=audio 8000 RTP/AVP 0
+			a=rtpmap:0 PCMU/8000
+
+		]]></send>
+	<send><![CDATA[
+
+			SIP/2.0 180 Ringing
+			[last_Via:]
+			[last_Call-ID:]
+			[last_From:]
+			[last_To:]
+			[last_CSeq:]
+			Content-Length: 0
+
+		]]></send>
+	<send retrans="500"><![CDATA[
+
+			SIP/2.0 200 OK
+			[last_Via:]
+			[last_Call-ID:]
+			[last_From:]
+			[last_To:];tag=[call_number]
+			[last_CSeq:]
+			Contact: <sip:736596371553211257@[local_ip]:[local_port];user=phone>
+			Content-Type: application/sdp
+			Content-Length: [len]
+
+			v=0
+			o=HuaweiSoftX3000 6644052 6644052 IN IP[local_ip_type] [local_ip]
+			s=Sip Call
+			c=IN IP[local_ip_type] [local_ip]
+			t=0 0
+			m=audio 8000 RTP/AVP 0
+			a=rtpmap:0 PCMU/8000
+
+		]]></send>
+	<recv request="ACK" rtd="true" crlf="true"/>
+	<recv request="BYE"/>
+	<send><![CDATA[
+
+			SIP/2.0 200 OK
+			[last_Via:]
+			[last_From:]
+			[last_To:]
+			[last_Call-ID:]
+			[last_CSeq:]
+			Content-Length: 0
+
+		]]></send>
+</scenario>
diff --git a/tests/channels/pjsip/allow_sending_180_after_183/disabled/test-config.yaml b/tests/channels/pjsip/allow_sending_180_after_183/disabled/test-config.yaml
new file mode 100644
index 0000000..e3fccfa
--- /dev/null
+++ b/tests/channels/pjsip/allow_sending_180_after_183/disabled/test-config.yaml
@@ -0,0 +1,27 @@
+testinfo:
+    summary: 'Test that Asterisk forward 180(SDP) as 183(SDP)'
+    description: |
+         'Asterisk is configured with "allow_sending_180_after_183=disable" where 180 most be changed to 183(SDP)'
+
+test-modules:
+    test-object:
+        config-section: test-object-config
+        typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+    memcheck-delay-stop: 7
+    fail-on-any: True
+    test-iterations:
+        -
+            scenarios:
+                - { 'key-args': {'scenario': 'A_PARTY.xml', '-i': '127.0.0.1', '-p': '5061', '-s': '3200000000'} }
+                - { 'key-args': {'scenario': 'B_PARTY.xml', '-i': '127.0.0.1', '-p': '5700'} }
+
+
+properties:
+    dependencies:
+        - sipp :
+             version : 'v3.0'
+        - asterisk : 'res_pjsip'
+    tags:
+        - pjsip
diff --git a/tests/channels/pjsip/allow_sending_180_after_183/enabled/configs/ast1/extensions.conf b/tests/channels/pjsip/allow_sending_180_after_183/enabled/configs/ast1/extensions.conf
new file mode 100644
index 0000000..ad7b155
--- /dev/null
+++ b/tests/channels/pjsip/allow_sending_180_after_183/enabled/configs/ast1/extensions.conf
@@ -0,0 +1,11 @@
+[general]
+static=yes
+writeprotect=yes
+autofallthrough=yes
+clearglobalvars=no
+priorityjumping=yes
+
+[globals]
+
+[default]
+exten => _X.,1,Dial(pjsip/sbc,180)
diff --git a/tests/channels/pjsip/allow_sending_180_after_183/enabled/configs/ast1/pjsip.conf b/tests/channels/pjsip/allow_sending_180_after_183/enabled/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..3738129
--- /dev/null
+++ b/tests/channels/pjsip/allow_sending_180_after_183/enabled/configs/ast1/pjsip.conf
@@ -0,0 +1,57 @@
+;--
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements start
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[general]
+sipdebug = yes
+
+[PEER_A]
+port = 5061
+
+[sbc]
+port = 5700
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+Non mapped elements end
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+--;
+
+
+[global]
+type = global
+debug = yes
+allow_sending_180_after_183=yes
+
+[transport-udp]
+type = transport
+protocol = udp
+bind = 0.0.0.0:5060
+
+[PEER_A]
+type = aor
+contact = sip:127.0.0.1:5061
+
+[PEER_A]
+type = identify
+endpoint = PEER_A
+match = 127.0.0.1
+
+[PEER_A]
+type = endpoint
+context = default
+allow = ulaw
+direct_media = no
+aors = PEER_A
+
+[sbc]
+type = aor
+contact = sip:127.0.0.1:5700
+
+[sbc]
+type = endpoint
+context = callcontrol
+allow = ulaw
+direct_media = no
+aors = sbc
+
diff --git a/tests/channels/pjsip/allow_sending_180_after_183/enabled/sipp/A_PARTY.xml b/tests/channels/pjsip/allow_sending_180_after_183/enabled/sipp/A_PARTY.xml
new file mode 100644
index 0000000..a3ce41b
--- /dev/null
+++ b/tests/channels/pjsip/allow_sending_180_after_183/enabled/sipp/A_PARTY.xml
@@ -0,0 +1,96 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<!-- This program is free software; you can redistribute it and/or      -->
+<!-- modify it under the terms of the GNU General Public License as     -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version.                    -->
+<!--                                                                    -->
+<!-- This program is distributed in the hope that it will be useful,    -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
+<!-- GNU General Public License for more details.                       -->
+<!--                                                                    -->
+<!-- You should have received a copy of the GNU General Public License  -->
+<!-- along with this program; if not, write to the                      -->
+<!-- Free Software Foundation, Inc.,                                    -->
+<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
+<!--                                                                    -->
+<!--                 Sipp 'uac' scenario with pcap (rtp) play           -->
+<!--                                                                    -->
+<scenario name="CONTENT_TYPE_PARAMS">
+	<!-- In client mode (sipp placing calls), the Call-ID MUST be		-->
+	<!-- generated by sipp. To do so, use [call_id] keyword.		-->
+	<send retrans="500"><![CDATA[
+
+			INVITE sip:[service]@uni-tel.dk SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: sipp <sip:test at uni-tel.dk>;tag=[call_number]
+			To: mpe <sip:[service]@uni-tel.dk:[remote_port]>
+			Call-ID: [call_id]
+			CSeq: 1 INVITE
+			Contact: sip:sipp@[local_ip]:[local_port]
+			Max-Forwards: 70
+			Subject: Performance Test
+			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:0 PCMU/8000
+
+		]]></send>
+	<recv response="100" optional="true"/>
+	<recv response="183">
+		<action>
+			<ereg regexp="application/sdp" search_in="hdr" header="Content-Type:" check_it="true" assign_to="1"/>
+		</action>
+	</recv>
+	<Reference variables="1"/>
+	<recv response="180">
+		<action>
+			<ereg regexp="application/sdp" search_in="hdr" header="Content-Type:" check_it="true" assign_to="2"/>
+		</action>
+	</recv>
+	<Reference variables="2"/>
+	<!-- By adding rrs="true" (Record Route Sets), the route sets		-->
+	<!-- are saved and used for following messages sent. Useful to test	-->
+	<!-- against stateful SIP proxies/B2BUAs.				-->
+	<recv response="200" rtd="true" crlf="true"/>
+	<!-- Packet lost can be simulated in any send/recv message by		-->
+	<!-- by adding the 'lost = "10"'. Value can be [1-100] percent.		-->
+	<send><![CDATA[
+
+			ACK sip:[service]@uni-tel.dk SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: sipp <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+			To: mpe <sip:[service]@uni-tel.dk>[peer_tag_param]
+			Call-ID: [call_id]
+			CSeq: 1 ACK
+			Contact: sip:sipp@[local_ip]:[local_port]
+			Max-Forwards: 70
+			Subject: Performance Test
+			Content-Length: 0
+
+		]]></send>
+	<pause milliseconds="1000"/>
+	<!-- The 'crlf' option inserts a blank line in the statistics report.	-->
+	<send retrans="500"><![CDATA[
+
+			BYE sip:[service]@uni-tel.dk SIP/2.0
+			Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+			From: sipp <sip:test@[local_ip]:[local_port]>;tag=[call_number]
+			To: mpe <sip:[service]@uni-tel.dk>[peer_tag_param]
+			Call-ID: [call_id]
+			CSeq: 2 BYE
+			Contact: sip:sipp@[local_ip]:[local_port]
+			Max-Forwards: 70
+			Subject: Performance Test
+			Content-Length: 0
+
+		]]></send>
+	<recv response="200" crlf="true"/>
+</scenario>
diff --git a/tests/channels/pjsip/allow_sending_180_after_183/enabled/sipp/B_PARTY.xml b/tests/channels/pjsip/allow_sending_180_after_183/enabled/sipp/B_PARTY.xml
new file mode 100644
index 0000000..89fc0ea
--- /dev/null
+++ b/tests/channels/pjsip/allow_sending_180_after_183/enabled/sipp/B_PARTY.xml
@@ -0,0 +1,87 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<scenario name="CONTENT_TYPE_PARAMS">
+	<User variables="dummy"/>
+	<recv request="INVITE" crlf="true" rrs="true">
+		<action>
+			<ereg regexp=".*sip:asterisk at .*" search_in="hdr" header="Contact:" check_it="true" assign_to="dummy"/>
+		</action>
+	</recv>
+	<Reference variables="dummy"/>
+	<send><![CDATA[
+
+			SIP/2.0 100 Trying
+			[last_Via:]
+			[last_Call-ID:]
+			[last_From:]
+			[last_To:]
+			[last_CSeq:]
+			Content-Length: 0
+
+		]]></send>
+	<send><![CDATA[
+
+			SIP/2.0 183 Session Progress
+			[last_Via:]
+			[last_Call-ID:]
+			[last_From:]
+			[last_To:]
+			[last_CSeq:]
+			Content-Type: application/sdp
+			Content-Length: [len]
+
+			v=0
+			o=HuaweiSoftX3000 6644052 6644052 IN IP[local_ip_type] [local_ip]
+			s=Sip Call
+			c=IN IP[local_ip_type] [local_ip]
+			t=0 0
+			m=audio 8000 RTP/AVP 0
+			a=rtpmap:0 PCMU/8000
+
+		]]></send>
+	<send><![CDATA[
+
+			SIP/2.0 180 Ringing
+			[last_Via:]
+			[last_Call-ID:]
+			[last_From:]
+			[last_To:]
+			[last_CSeq:]
+			Content-Length: 0
+
+			]]></send>
+	<send retrans="500"><![CDATA[
+
+			SIP/2.0 200 OK
+			[last_Via:]
+			[last_Call-ID:]
+			[last_From:]
+			[last_To:];tag=[call_number]
+			[last_CSeq:]
+			Contact: <sip:736596371553211257@[local_ip]:[local_port];user=phone>
+			Content-Type: application/sdp
+			Content-Length: [len]
+
+			v=0
+			o=HuaweiSoftX3000 6644052 6644052 IN IP[local_ip_type] [local_ip]
+			s=Sip Call
+			c=IN IP[local_ip_type] [local_ip]
+			t=0 0
+			m=audio 8000 RTP/AVP 0
+			a=rtpmap:0 PCMU/8000
+
+		]]></send>
+	<recv request="ACK" rtd="true" crlf="true"/>
+	<recv request="BYE"/>
+	<send><![CDATA[
+
+			SIP/2.0 200 OK
+			[last_Via:]
+			[last_From:]
+			[last_To:]
+			[last_Call-ID:]
+			[last_CSeq:]
+			Content-Length: 0
+
+		]]></send>
+</scenario>
diff --git a/tests/channels/pjsip/allow_sending_180_after_183/enabled/test-config.yaml b/tests/channels/pjsip/allow_sending_180_after_183/enabled/test-config.yaml
new file mode 100644
index 0000000..5be01dc
--- /dev/null
+++ b/tests/channels/pjsip/allow_sending_180_after_183/enabled/test-config.yaml
@@ -0,0 +1,27 @@
+testinfo:
+    summary: 'Test that Asterisk forward 180(SDP) as 180(SDP)'
+    description: |
+         'Asterisk is configured with "allow_sending_180_after_183=enabled" where 180 most be remain as 180(SDP)'
+
+test-modules:
+    test-object:
+        config-section: test-object-config
+        typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+    memcheck-delay-stop: 7
+    fail-on-any: True
+    test-iterations:
+        -
+            scenarios:
+                - { 'key-args': {'scenario': 'A_PARTY.xml', '-i': '127.0.0.1', '-p': '5061', '-s': '3200000000'} }
+                - { 'key-args': {'scenario': 'B_PARTY.xml', '-i': '127.0.0.1', '-p': '5700'} }
+
+
+properties:
+    dependencies:
+        - sipp :
+             version : 'v3.0'
+        - asterisk : 'res_pjsip'
+    tags:
+        - pjsip
diff --git a/tests/channels/pjsip/allow_sending_180_after_183/tests.yaml b/tests/channels/pjsip/allow_sending_180_after_183/tests.yaml
new file mode 100644
index 0000000..783cb68
--- /dev/null
+++ b/tests/channels/pjsip/allow_sending_180_after_183/tests.yaml
@@ -0,0 +1,3 @@
+tests:
+    - test: 'disabled'
+    - test: 'enabled'
diff --git a/tests/channels/pjsip/tests.yaml b/tests/channels/pjsip/tests.yaml
index 1b4fad8..e585356 100644
--- a/tests/channels/pjsip/tests.yaml
+++ b/tests/channels/pjsip/tests.yaml
@@ -1,5 +1,6 @@
 # Enter tests here in the order they should be considered for execution:
 tests:
+    - dir: 'allow_sending_180_after_183'
     - dir: 'ami'
     - dir: 'auth'
     - dir: 'basic_calls'

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Gerrit-Project: testsuite
Gerrit-Branch: 19
Gerrit-Change-Id: Ie0ba7dd79f881110fc7d3f000e72dfc52dd7edf3
Gerrit-Change-Number: 18447
Gerrit-PatchSet: 1
Gerrit-Owner: Mark Petersen <asterisk.org at zombie.dk>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-CC: Mark Petersen <bugs.digium.com at zombie.dk>
Gerrit-MessageType: merged
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