[asterisk-commits] kharwell: testsuite/asterisk/trunk r4204 - in /asterisk/trunk/tests/channels/...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Sep 18 09:27:11 CDT 2013
Author: kharwell
Date: Wed Sep 18 09:27:02 2013
New Revision: 4204
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=4204
Log:
Testsuite: pjsip call forwarding tests
Call forwarding testsuite tests for the new sip channel driver. Various
tests to make sure the diversion header gets appropriately added during
call forwarding.
Review: https://reviewboard.asterisk.org/r/2837/
Added:
asterisk/trunk/tests/channels/pjsip/diversion/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/configs/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/configs/ast1/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/configs/ast1/pjsip.conf (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/sipp/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/sipp/user1.xml (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/sipp/user2.xml (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/sipp/user3.xml (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/test-config.yaml (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/configs/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/configs/ast1/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/configs/ast1/pjsip.conf (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/sipp/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/sipp/user1.xml (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/sipp/user2.xml (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/sipp/user3.xml (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/test-config.yaml (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/configs/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/configs/ast1/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/configs/ast1/pjsip.conf (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/sipp/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/sipp/user1.xml (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/sipp/user2.xml (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/test-config.yaml (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_response/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_response/configs/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_response/configs/ast1/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_response/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_response/configs/ast1/pjsip.conf (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_response/sipp/
asterisk/trunk/tests/channels/pjsip/diversion/diversion_response/sipp/user1.xml (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_response/sipp/user2.xml (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_response/sipp/user3.xml (with props)
asterisk/trunk/tests/channels/pjsip/diversion/diversion_response/test-config.yaml (with props)
asterisk/trunk/tests/channels/pjsip/diversion/tests.yaml (with props)
Modified:
asterisk/trunk/tests/channels/pjsip/tests.yaml
Added: asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/configs/ast1/extensions.conf?view=auto&rev=4204
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/configs/ast1/extensions.conf Wed Sep 18 09:27:02 2013
@@ -1,0 +1,7 @@
+[general]
+
+[default]
+
+exten => user1,1,Dial(PJSIP/user1)
+exten => user2,1,Dial(PJSIP/user2)
+exten => user3,1,Dial(PJSIP/user3)
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Added: asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/configs/ast1/pjsip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/configs/ast1/pjsip.conf?view=auto&rev=4204
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/configs/ast1/pjsip.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/configs/ast1/pjsip.conf Wed Sep 18 09:27:02 2013
@@ -1,0 +1,40 @@
+[local]
+type=transport
+protocol=udp
+bind=0.0.0.0
+
+[user1-aors]
+type=aor
+contact=sip:user1 at 127.0.0.1:5061
+
+[user1]
+type=endpoint
+context=default
+aors=user1-aors
+direct_media=no
+disallow=all
+allow=ulaw
+
+[user2-aors]
+type=aor
+contact=sip:user2 at 127.0.0.1:5062
+
+[user2]
+type=endpoint
+context=default
+aors=user2-aors
+direct_media=no
+disallow=all
+allow=ulaw
+
+[user3-aors]
+type=aor
+contact=sip:user3 at 127.0.0.1:5063
+
+[user3]
+type=endpoint
+context=default
+aors=user3-aors
+direct_media=no
+disallow=all
+allow=ulaw
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Added: asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/sipp/user1.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/sipp/user1.xml?view=auto&rev=4204
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/sipp/user1.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/sipp/user1.xml Wed Sep 18 09:27:02 2013
@@ -1,0 +1,68 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="send an invite to user2">
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:user2@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: user1 <sip:user1@[local_ip]:[local_port]>;tag=[call_number]
+ To: user2 <sip:user2@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: [cseq] INVITE
+ Contact: sip:user1@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0 101
+ a=sendrecv
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true" />
+
+ <recv response="181" optional="true" />
+
+ <recv response="180" optional="true" />
+
+ <recv response="183" optional="true" />
+
+ <recv response="200" />
+
+ <send>
+ <![CDATA[
+ ACK sip:user3@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:user1@[local_ip]>;tag=[call_number]
+ To: <sip:user3@[remote_ip]:[remote_port]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [call_id]
+ Max-Forwards: 70
+ Content-Length: 0
+
+ ]]>
+ </send>
+ <pause milliseconds="3000"/>
+ <send retrans="500">
+ <![CDATA[
+ BYE sip:user3@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:user1@[local_ip]>;tag=[call_number]
+ To: <sip:user3@[remote_ip]:[remote_port]>[peer_tag_param]
+ CSeq: [cseq] BYE
+ Call-ID: [call_id]
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true" />
+</scenario>
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Added: asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/sipp/user2.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/sipp/user2.xml?view=auto&rev=4204
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/sipp/user2.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/sipp/user2.xml Wed Sep 18 09:27:02 2013
@@ -1,0 +1,20 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="send forwarding response back">
+ <recv request="INVITE" crlf="true" />
+
+ <send>
+ <![CDATA[
+ SIP/2.0 302 Moved Temporarily
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:user3@[local_ip]>
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv request="ACK"/>
+</scenario>
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Added: asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/sipp/user3.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/sipp/user3.xml?view=auto&rev=4204
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/sipp/user3.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/sipp/user3.xml Wed Sep 18 09:27:02 2013
@@ -1,0 +1,80 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="receives forwarded call from user2">
+ <recv request="INVITE" crlf="true">
+ <action>
+ <!-- Check that the Diversion header is present. -->
+ <ereg regexp=".*" search_in="hdr" header="Diversion:" check_it="true" assign_to="1" />
+ <log message="Received INVITE with Diversion header: [$1]." />
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:user3@[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:user3@[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:user3@[local_ip]:[local_port];transport=[transport]>
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0 101
+ a=sendrecv
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+
+ ]]>
+ </send>
+
+ <recv request="ACK"/>
+
+ <recv request="BYE" crlf="true" />
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Type: application/sdp
+ Content-Length: 0
+
+ ]]>
+ </send>
+</scenario>
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Added: asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/test-config.yaml?view=auto&rev=4204
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_basic/test-config.yaml Wed Sep 18 09:27:02 2013
@@ -1,0 +1,25 @@
+testinfo:
+ summary: 'Test to make sure an appropriate diversion header gets added when
+ a call is forwarded.'
+ description: |
+ 'user1 calls user2 which is set to forward calls to user3.'
+
+properties:
+ minversion: '12'
+ dependencies:
+ - app : 'sipp'
+ tags:
+ - SIP
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': { 'scenario':'user3.xml', '-p':'5063' } }
+ - { 'key-args': { 'scenario':'user2.xml', '-p':'5062' } }
+ - { 'key-args': { 'scenario':'user1.xml', '-p':'5061' } }
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Added: asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/configs/ast1/extensions.conf?view=auto&rev=4204
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/configs/ast1/extensions.conf Wed Sep 18 09:27:02 2013
@@ -1,0 +1,7 @@
+[general]
+
+[default]
+
+exten => user1,1,Dial(PJSIP/user1)
+exten => user2,1,Dial(PJSIP/user2)
+exten => user3,1,Dial(PJSIP/user3)
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Added: asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/configs/ast1/pjsip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/configs/ast1/pjsip.conf?view=auto&rev=4204
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/configs/ast1/pjsip.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/configs/ast1/pjsip.conf Wed Sep 18 09:27:02 2013
@@ -1,0 +1,46 @@
+[local]
+type=transport
+protocol=udp
+bind=0.0.0.0
+
+[user1-aors]
+type=aor
+contact=sip:user1 at 127.0.0.1:5061
+
+[user1]
+type=endpoint
+context=default
+aors=user1-aors
+direct_media=no
+disallow=all
+allow=ulaw
+callerid=khufu <4444>
+send_pai=yes
+
+[user2-aors]
+type=aor
+contact=sip:user2 at 127.0.0.1:5062
+
+[user2]
+type=endpoint
+context=default
+aors=user2-aors
+direct_media=no
+disallow=all
+allow=ulaw
+callerid=amenhotep <1234>
+send_pai=yes
+trust_id_outbound=yes
+
+[user3-aors]
+type=aor
+contact=sip:user3 at 127.0.0.1:5063
+
+[user3]
+type=endpoint
+context=default
+aors=user3-aors
+direct_media=no
+disallow=all
+allow=ulaw
+send_pai=yes
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Added: asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/sipp/user1.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/sipp/user1.xml?view=auto&rev=4204
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/sipp/user1.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/sipp/user1.xml Wed Sep 18 09:27:02 2013
@@ -1,0 +1,68 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="send an invite to user2">
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:user2@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: user1 <sip:user1@[local_ip]:[local_port]>;tag=[call_number]
+ To: user2 <sip:user2@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: [cseq] INVITE
+ Contact: sip:user1@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0 101
+ a=sendrecv
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true" />
+
+ <recv response="181" optional="true" />
+
+ <recv response="180" optional="true" />
+
+ <recv response="183" optional="true" />
+
+ <recv response="200" />
+
+ <send>
+ <![CDATA[
+ ACK sip:user3@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:user1@[local_ip]>;tag=[call_number]
+ To: <sip:user3@[remote_ip]:[remote_port]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [call_id]
+ Max-Forwards: 70
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+ BYE sip:user3@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:user1@[local_ip]>;tag=[call_number]
+ To: <sip:user3@[remote_ip]:[remote_port]>[peer_tag_param]
+ CSeq: [cseq] BYE
+ Call-ID: [call_id]
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true" />
+</scenario>
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svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/sipp/user2.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/sipp/user2.xml?view=auto&rev=4204
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/sipp/user2.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/sipp/user2.xml Wed Sep 18 09:27:02 2013
@@ -1,0 +1,20 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="send forwarding response back">
+ <recv request="INVITE" crlf="true" />
+
+ <send>
+ <![CDATA[
+ SIP/2.0 302 Moved Temporarily
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:user3@[local_ip]>
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv request="ACK"/>
+</scenario>
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Added: asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/sipp/user3.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/sipp/user3.xml?view=auto&rev=4204
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/sipp/user3.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/sipp/user3.xml Wed Sep 18 09:27:02 2013
@@ -1,0 +1,80 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="receives forwarded call from user2">
+ <recv request="INVITE" crlf="true">
+ <action>
+ <!-- Check that the Diversion header is present and contains the correct caller id. -->
+ <ereg regexp="user2" search_in="hdr" header="Diversion:" check_it="true" assign_to="1" />
+ <log message="Received INVITE with Diversion header: [$1]." />
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:user3@[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:user3@[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:user3@[local_ip]:[local_port];transport=[transport]>
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0 101
+ a=sendrecv
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+
+ ]]>
+ </send>
+
+ <recv request="ACK"/>
+
+ <recv request="BYE" crlf="true" />
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Type: application/sdp
+ Content-Length: 0
+
+ ]]>
+ </send>
+</scenario>
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--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_caller_id/test-config.yaml Wed Sep 18 09:27:02 2013
@@ -1,0 +1,25 @@
+testinfo:
+ summary: 'Test to make sure an appropriate diversion header gets added when
+ a call is forwarded.'
+ description: |
+ 'user1 calls user2 which is set to forward calls to user3.'
+
+properties:
+ minversion: '12'
+ dependencies:
+ - app : 'sipp'
+ tags:
+ - SIP
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': { 'scenario':'user3.xml', '-p':'5063' } }
+ - { 'key-args': { 'scenario':'user2.xml', '-p':'5062' } }
+ - { 'key-args': { 'scenario':'user1.xml', '-p':'5061' } }
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Added: asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/configs/ast1/extensions.conf
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==============================================================================
--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/configs/ast1/extensions.conf Wed Sep 18 09:27:02 2013
@@ -1,0 +1,7 @@
+[general]
+
+[default]
+
+exten => user1,1,Dial(PJSIP/user1)
+exten => user2,1,Dial(PJSIP/user2)
+exten => user3,1,Dial(PJSIP/user3)
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Added: asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/configs/ast1/pjsip.conf
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==============================================================================
--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/configs/ast1/pjsip.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/configs/ast1/pjsip.conf Wed Sep 18 09:27:02 2013
@@ -1,0 +1,40 @@
+[local]
+type=transport
+protocol=udp
+bind=0.0.0.0
+
+[user1-aors]
+type=aor
+contact=sip:user1 at 127.0.0.1:5061
+
+[user1]
+type=endpoint
+context=default
+aors=user1-aors
+direct_media=no
+disallow=all
+allow=ulaw
+
+[user2-aors]
+type=aor
+contact=sip:user2 at 127.0.0.1:5062
+
+[user2]
+type=endpoint
+context=default
+aors=user2-aors
+direct_media=no
+disallow=all
+allow=ulaw
+
+[user3-aors]
+type=aor
+contact=sip:user3 at 127.0.0.1:5063
+
+[user3]
+type=endpoint
+context=default
+aors=user3-aors
+direct_media=no
+disallow=all
+allow=ulaw
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--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/sipp/user1.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/sipp/user1.xml Wed Sep 18 09:27:02 2013
@@ -1,0 +1,69 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="send an invite to user2">
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:user2@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: user1 <sip:user1@[local_ip]:[local_port]>;tag=[call_number]
+ To: user2 <sip:user2@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: [cseq] INVITE
+ Contact: sip:user1@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Diversion: <sip:amenhotep at 127.0.0.1>;reason=unconditional
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0 101
+ a=sendrecv
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true" />
+
+ <recv response="181" optional="true" />
+
+ <recv response="180" optional="true" />
+
+ <recv response="183" optional="true" />
+
+ <recv response="200" />
+
+ <send>
+ <![CDATA[
+ ACK sip:user2@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:user1@[local_ip]>;tag=[call_number]
+ To: <sip:user2@[remote_ip]:[remote_port]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [call_id]
+ Max-Forwards: 70
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+ BYE sip:user2@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:user1@[local_ip]>;tag=[call_number]
+ To: <sip:user2@[remote_ip]:[remote_port]>[peer_tag_param]
+ CSeq: [cseq] BYE
+ Call-ID: [call_id]
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true" />
+</scenario>
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==============================================================================
--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/sipp/user2.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/sipp/user2.xml Wed Sep 18 09:27:02 2013
@@ -1,0 +1,80 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="receives invite with diversion info">
+ <recv request="INVITE" crlf="true">
+ <action>
+ <!-- Check that the Diversion header is present and contains the correct name. -->
+ <ereg regexp="amenhotep" search_in="hdr" header="Diversion:" check_it="true" assign_to="1" />
+ <log message="Received INVITE with Diversion header: [$1]." />
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:user2@[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:user2@[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:user2@[local_ip]:[local_port];transport=[transport]>
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0 101
+ a=sendrecv
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+
+ ]]>
+ </send>
+
+ <recv request="ACK"/>
+
+ <recv request="BYE" crlf="true" />
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Type: application/sdp
+ Content-Length: 0
+
+ ]]>
+ </send>
+</scenario>
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==============================================================================
--- asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/pjsip/diversion/diversion_request/test-config.yaml Wed Sep 18 09:27:02 2013
@@ -1,0 +1,24 @@
+testinfo:
+ summary: 'Test to make sure the diversion header on an invite gets
+ propagated correctly.'
+ description: |
+ 'user1 calls user2 with an invite containing a diversion header.'
+
+properties:
+ minversion: '12'
+ dependencies:
+ - app : 'sipp'
+ tags:
+ - SIP
+
+test-modules:
+ test-object:
[... 415 lines stripped ...]
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