[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'
--
To view, visit https://gerrit.asterisk.org/c/testsuite/+/18447
To unsubscribe, or for help writing mail filters, visit https://gerrit.asterisk.org/settings
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-commits/attachments/20220426/af128857/attachment-0001.html>
More information about the asterisk-commits
mailing list