[asterisk-commits] russell: testsuite/asterisk/trunk r1612 - in /asterisk/trunk: ./ lib/python/a...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Jun 2 17:20:59 CDT 2011
Author: russell
Date: Thu Jun 2 17:20:54 2011
New Revision: 1612
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=1612
Log:
Add 4 new tests related to SIP MESSAGE handling.
message_diabled - check response to MESSAGE when out of call messages
are disabled
message_unauth - Pass through MESSAGE without authentication
message_auth - Pass through MESSAGE with authentication
message_from_call - Send a MESSAGE to another party from dialplan being
executed as the result from a normal call
Added:
asterisk/trunk/tests/channels/SIP/message_auth/
asterisk/trunk/tests/channels/SIP/message_auth/configs/
asterisk/trunk/tests/channels/SIP/message_auth/configs/ast1/
asterisk/trunk/tests/channels/SIP/message_auth/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/SIP/message_auth/configs/ast1/sip.conf (with props)
asterisk/trunk/tests/channels/SIP/message_auth/configs/sip.conf (with props)
asterisk/trunk/tests/channels/SIP/message_auth/run-test (with props)
asterisk/trunk/tests/channels/SIP/message_auth/sipp/
asterisk/trunk/tests/channels/SIP/message_auth/sipp/message.xml (with props)
asterisk/trunk/tests/channels/SIP/message_auth/sipp/message_recv.xml (with props)
asterisk/trunk/tests/channels/SIP/message_auth/test-config.yaml (with props)
asterisk/trunk/tests/channels/SIP/message_disabled/
asterisk/trunk/tests/channels/SIP/message_disabled/configs/
asterisk/trunk/tests/channels/SIP/message_disabled/configs/ast1/
asterisk/trunk/tests/channels/SIP/message_disabled/configs/ast1/sip.conf (with props)
asterisk/trunk/tests/channels/SIP/message_disabled/run-test (with props)
asterisk/trunk/tests/channels/SIP/message_disabled/sipp/
asterisk/trunk/tests/channels/SIP/message_disabled/sipp/message.xml (with props)
asterisk/trunk/tests/channels/SIP/message_disabled/test-config.yaml (with props)
asterisk/trunk/tests/channels/SIP/message_from_call/
asterisk/trunk/tests/channels/SIP/message_from_call/configs/
asterisk/trunk/tests/channels/SIP/message_from_call/configs/ast1/
asterisk/trunk/tests/channels/SIP/message_from_call/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/SIP/message_from_call/configs/ast1/sip.conf (with props)
asterisk/trunk/tests/channels/SIP/message_from_call/run-test (with props)
asterisk/trunk/tests/channels/SIP/message_from_call/sipp/
asterisk/trunk/tests/channels/SIP/message_from_call/sipp/call.xml (with props)
asterisk/trunk/tests/channels/SIP/message_from_call/sipp/message_recv.xml (with props)
asterisk/trunk/tests/channels/SIP/message_from_call/test-config.yaml (with props)
asterisk/trunk/tests/channels/SIP/message_unauth/
asterisk/trunk/tests/channels/SIP/message_unauth/configs/
asterisk/trunk/tests/channels/SIP/message_unauth/configs/ast1/
asterisk/trunk/tests/channels/SIP/message_unauth/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/SIP/message_unauth/configs/ast1/sip.conf (with props)
asterisk/trunk/tests/channels/SIP/message_unauth/run-test (with props)
asterisk/trunk/tests/channels/SIP/message_unauth/sipp/
asterisk/trunk/tests/channels/SIP/message_unauth/sipp/message.xml (with props)
asterisk/trunk/tests/channels/SIP/message_unauth/sipp/message_recv.xml (with props)
asterisk/trunk/tests/channels/SIP/message_unauth/test-config.yaml (with props)
Modified:
asterisk/trunk/README.txt
asterisk/trunk/lib/python/asterisk/sipp.py
asterisk/trunk/tests/channels/SIP/tests.yaml
Modified: asterisk/trunk/README.txt
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/README.txt?view=diff&rev=1612&r1=1611&r2=1612
==============================================================================
--- asterisk/trunk/README.txt (original)
+++ asterisk/trunk/README.txt Thu Jun 2 17:20:54 2011
@@ -3,7 +3,7 @@
=== Asterisk Test Suite ===
=== ===
=== http://www.asterisk.org/ ===
-=== Copyright (C) 2010, Digium, Inc. ===
+=== Copyright (C) 2010 - 2011, Digium, Inc. ===
=== ===
================================================================================
Modified: asterisk/trunk/lib/python/asterisk/sipp.py
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/lib/python/asterisk/sipp.py?view=diff&rev=1612&r1=1611&r2=1612
==============================================================================
--- asterisk/trunk/lib/python/asterisk/sipp.py (original)
+++ asterisk/trunk/lib/python/asterisk/sipp.py Thu Jun 2 17:20:54 2011
@@ -91,6 +91,7 @@
sipp_args.extend([ key, val ])
print "Executing SIPp scenario: %s" % scenario['scenario']
+ print sipp_args
return subprocess.Popen(sipp_args,
stdout=subprocess.PIPE, stderr=subprocess.PIPE)
Added: asterisk/trunk/tests/channels/SIP/message_auth/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_auth/configs/ast1/extensions.conf?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_auth/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/SIP/message_auth/configs/ast1/extensions.conf Thu Jun 2 17:20:54 2011
@@ -1,0 +1,10 @@
+[default]
+
+exten => _.,1,NoOp(ERROR: THIS SHOULD NOT BE EXECUTED)
+
+[not_default]
+
+exten => _.,1,NoOp()
+
+exten => user,1,Verbose(1,Got a message from SIPp)
+exten => user,n,MessageSend(sip:user2)
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Added: asterisk/trunk/tests/channels/SIP/message_auth/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_auth/configs/ast1/sip.conf?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_auth/configs/ast1/sip.conf (added)
+++ asterisk/trunk/tests/channels/SIP/message_auth/configs/ast1/sip.conf Thu Jun 2 17:20:54 2011
@@ -1,0 +1,20 @@
+[general]
+
+accept_outofcall_message = yes
+auth_message_requests = yes
+
+[user]
+type=peer
+host=127.0.0.1
+port=5061
+context=not_default
+secret=blah
+
+[user2]
+type=peer
+host=127.0.0.1
+port=5062
+context=not_default
+username=user2
+secret=blah
+
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Added: asterisk/trunk/tests/channels/SIP/message_auth/configs/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_auth/configs/sip.conf?view=auto&rev=1612
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(empty)
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Added: asterisk/trunk/tests/channels/SIP/message_auth/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_auth/run-test?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_auth/run-test (added)
+++ asterisk/trunk/tests/channels/SIP/message_auth/run-test Thu Jun 2 17:20:54 2011
@@ -1,0 +1,44 @@
+#!/usr/bin/env python
+'''
+Copyright (C) 2010, Digium, Inc.
+Russell Bryant <russell at digium.com>
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+'''
+
+import sys
+import os
+
+sys.path.append("lib/python")
+from asterisk.sipp import SIPpTest
+
+
+WORKING_DIR = "/tmp/asterisk-testsuite/sip/message_auth"
+TEST_DIR = os.path.dirname(os.path.realpath(__file__))
+
+SIPP_SCENARIOS = [
+ {
+ 'scenario' : 'message_recv.xml',
+ '-p' : '5062'
+ },
+ {
+ 'scenario' : 'message.xml',
+ '-p' : '5061'
+ }
+]
+
+
+def main(argv=None):
+ if argv is None:
+ argv = sys.argv
+
+ test = SIPpTest(WORKING_DIR, TEST_DIR, SIPP_SCENARIOS)
+ return test.run()
+
+
+if __name__ == "__main__":
+ sys.exit(main())
+
+
+# vim:sw=4:ts=4:expandtab:textwidth=79
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Added: asterisk/trunk/tests/channels/SIP/message_auth/sipp/message.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_auth/sipp/message.xml?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_auth/sipp/message.xml (added)
+++ asterisk/trunk/tests/channels/SIP/message_auth/sipp/message.xml Thu Jun 2 17:20:54 2011
@@ -1,0 +1,48 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="MESSAGE with auth">
+ <!-- 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[
+ MESSAGE sip:user@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: user <sip:user@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: <sip:user@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 MESSAGE
+ Contact: sip:user@[local_ip]:[local_port]
+ Max-Forwards: 20
+ Expires: 3600
+ Content-Type: text/plain
+ Content-Length: 18
+
+ Watson, come here.
+
+ ]]>
+ </send>
+
+ <recv response="401" auth="true" rtd="true"/>
+
+ <send retrans="500">
+ <![CDATA[
+ MESSAGE sip:user@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ Max-Forwards: 20
+ From: user <sip:user@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: <sip:user@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 2 MESSAGE
+ Contact: sip:user@[local_ip]:[local_port]
+ Expires: 3600
+ [authentication username=user password=blah]
+ Content-Type: text/plain
+ Content-Length: 18
+
+ Watson, come here.
+
+ ]]>
+ </send>
+
+ <recv response="202" rtd="true" />
+</scenario>
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Added: asterisk/trunk/tests/channels/SIP/message_auth/sipp/message_recv.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_auth/sipp/message_recv.xml?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_auth/sipp/message_recv.xml (added)
+++ asterisk/trunk/tests/channels/SIP/message_auth/sipp/message_recv.xml Thu Jun 2 17:20:54 2011
@@ -1,0 +1,37 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="Basic MESSAGE send and receive">
+ <recv request="MESSAGE" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 401 Unauthorized
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ WWW-Authenticate: digest realm="test",stale=true
+
+ ]]>
+ </send>
+
+ <recv request="MESSAGE" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 202 Accepted
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
+ Supported: replaces, timer
+ Content-Length: 0
+
+ ]]>
+ </send>
+</scenario>
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Added: asterisk/trunk/tests/channels/SIP/message_auth/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_auth/test-config.yaml?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_auth/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/SIP/message_auth/test-config.yaml Thu Jun 2 17:20:54 2011
@@ -1,0 +1,11 @@
+testinfo:
+ summary: 'Test inbound and outbound unauthenticated MESSAGE'
+ description: |
+ 'Send Asterisk a MESSAGE and wait for Asterisk to send it back.'
+
+properties:
+ minversion: '1.10'
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ feature : 'TLS'
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Added: asterisk/trunk/tests/channels/SIP/message_disabled/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_disabled/configs/ast1/sip.conf?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_disabled/configs/ast1/sip.conf (added)
+++ asterisk/trunk/tests/channels/SIP/message_disabled/configs/ast1/sip.conf Thu Jun 2 17:20:54 2011
@@ -1,0 +1,3 @@
+[general]
+
+accept_outofcall_message = no
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Added: asterisk/trunk/tests/channels/SIP/message_disabled/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_disabled/run-test?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_disabled/run-test (added)
+++ asterisk/trunk/tests/channels/SIP/message_disabled/run-test Thu Jun 2 17:20:54 2011
@@ -1,0 +1,39 @@
+#!/usr/bin/env python
+'''
+Copyright (C) 2010, Digium, Inc.
+Russell Bryant <russell at digium.com>
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+'''
+
+import sys
+import os
+
+sys.path.append("lib/python")
+from asterisk.sipp import SIPpTest
+
+
+WORKING_DIR = "/tmp/asterisk-testsuite/sip/message_disabled"
+TEST_DIR = os.path.dirname(os.path.realpath(__file__))
+
+SIPP_SCENARIOS = [
+ {
+ 'scenario' : 'message.xml',
+ }
+]
+
+
+def main(argv=None):
+ if argv is None:
+ argv = sys.argv
+
+ test = SIPpTest(WORKING_DIR, TEST_DIR, SIPP_SCENARIOS)
+ return test.run()
+
+
+if __name__ == "__main__":
+ sys.exit(main())
+
+
+# vim:sw=4:ts=4:expandtab:textwidth=79
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--- asterisk/trunk/tests/channels/SIP/message_disabled/sipp/message.xml (added)
+++ asterisk/trunk/tests/channels/SIP/message_disabled/sipp/message.xml Thu Jun 2 17:20:54 2011
@@ -1,0 +1,27 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="Basic MESSAGE send and receive">
+ <!-- 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[
+
+ MESSAGE sip:user@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: user <sip:user@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: <sip:user@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 MESSAGE
+ Contact: sip:user@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Expires: 3600
+ Content-Type: text/plain
+ Content-Length: 18
+
+ Watson, come here.
+
+ ]]>
+ </send>
+
+ <recv response="405" rtd="true" />
+</scenario>
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Added: asterisk/trunk/tests/channels/SIP/message_disabled/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_disabled/test-config.yaml?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_disabled/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/SIP/message_disabled/test-config.yaml Thu Jun 2 17:20:54 2011
@@ -1,0 +1,9 @@
+testinfo:
+ summary: 'Test response to MESSAGE when out-of-call MESSAGE is disabled (or not supported)'
+ description: |
+ 'Send Asterisk a MESSAGE and expect that it gets denied'
+
+properties:
+ minversion: '1.10'
+ dependencies:
+ - app : 'sipp'
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Added: asterisk/trunk/tests/channels/SIP/message_from_call/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_from_call/configs/ast1/extensions.conf?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_from_call/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/SIP/message_from_call/configs/ast1/extensions.conf Thu Jun 2 17:20:54 2011
@@ -1,0 +1,8 @@
+[default]
+
+exten => _.,1,NoOp()
+
+exten => test,1,Answer()
+exten => test,n,Set(MESSAGE(body)=I like waffles.)
+exten => test,n,MessageSend(sip:user at 127.0.0.1:5062)
+exten => test,n,Wait(30) ; SIPp will hang up.
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--- asterisk/trunk/tests/channels/SIP/message_from_call/configs/ast1/sip.conf (added)
+++ asterisk/trunk/tests/channels/SIP/message_from_call/configs/ast1/sip.conf Thu Jun 2 17:20:54 2011
@@ -1,0 +1,3 @@
+[general]
+
+allowguest=yes
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Added: asterisk/trunk/tests/channels/SIP/message_from_call/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_from_call/run-test?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_from_call/run-test (added)
+++ asterisk/trunk/tests/channels/SIP/message_from_call/run-test Thu Jun 2 17:20:54 2011
@@ -1,0 +1,44 @@
+#!/usr/bin/env python
+'''
+Copyright (C) 2010, Digium, Inc.
+Russell Bryant <russell at digium.com>
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+'''
+
+import sys
+import os
+
+sys.path.append("lib/python")
+from asterisk.sipp import SIPpTest
+
+
+WORKING_DIR = "/tmp/asterisk-testsuite/sip/message_from_call"
+TEST_DIR = os.path.dirname(os.path.realpath(__file__))
+
+SIPP_SCENARIOS = [
+ {
+ 'scenario' : 'message_recv.xml',
+ '-p' : '5062'
+ },
+ {
+ 'scenario' : 'call.xml',
+ '-p' : '5061'
+ }
+]
+
+
+def main(argv=None):
+ if argv is None:
+ argv = sys.argv
+
+ test = SIPpTest(WORKING_DIR, TEST_DIR, SIPP_SCENARIOS)
+ return test.run()
+
+
+if __name__ == "__main__":
+ sys.exit(main())
+
+
+# vim:sw=4:ts=4:expandtab:textwidth=79
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Added: asterisk/trunk/tests/channels/SIP/message_from_call/sipp/call.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_from_call/sipp/call.xml?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_from_call/sipp/call.xml (added)
+++ asterisk/trunk/tests/channels/SIP/message_from_call/sipp/call.xml Thu Jun 2 17:20:54 2011
@@ -1,0 +1,95 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Channel Test">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:wienerschnitzel@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ User-Agent: Channel Param 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[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0 101
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ m=video 6002 RTP/AVP 34
+ a=rtpmap:34 H263/90000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:wienerschnitzel@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- Pause long enough for our test to run -->
+ <pause milliseconds="4000" />
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:wienerschnitzel@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true">
+ </recv>
+
+ <!-- definition of the response time repartition table (unit is ms) -->
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <!-- definition of the call length repartition table (unit is ms) -->
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
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URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_from_call/sipp/message_recv.xml?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_from_call/sipp/message_recv.xml (added)
+++ asterisk/trunk/tests/channels/SIP/message_from_call/sipp/message_recv.xml Thu Jun 2 17:20:54 2011
@@ -1,0 +1,21 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="Basic MESSAGE send and receive">
+ <recv request="MESSAGE" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 202 Accepted
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
+ Supported: replaces, timer
+ Content-Length: 0
+
+ ]]>
+ </send>
+</scenario>
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Added: asterisk/trunk/tests/channels/SIP/message_from_call/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_from_call/test-config.yaml?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_from_call/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/SIP/message_from_call/test-config.yaml Thu Jun 2 17:20:54 2011
@@ -1,0 +1,10 @@
+testinfo:
+ summary: 'Test sending a MESSAGE while processing a call.'
+ description: |
+ 'Set up a call with Asterisk and send an out-of-call SIP MESSAGE
+ from the dialplan processing that call.'
+
+properties:
+ minversion: '1.10'
+ dependencies:
+ - app : 'sipp'
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Added: asterisk/trunk/tests/channels/SIP/message_unauth/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_unauth/configs/ast1/extensions.conf?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_unauth/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/SIP/message_unauth/configs/ast1/extensions.conf Thu Jun 2 17:20:54 2011
@@ -1,0 +1,9 @@
+[default]
+
+exten => _.,1,NoOp()
+
+exten => user,1,Verbose(1,Got a message from SIPp)
+exten => user,n,Verbose(1,To: ${MESSAGE(to)})
+exten => user,n,Verbose(1,From: ${MESSAGE(from)})
+exten => user,n,Verbose(1,Body: ${MESSAGE(body)})
+exten => user,n,MessageSend(sip:user at 127.0.0.1:5062)
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URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_unauth/configs/ast1/sip.conf?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_unauth/configs/ast1/sip.conf (added)
+++ asterisk/trunk/tests/channels/SIP/message_unauth/configs/ast1/sip.conf Thu Jun 2 17:20:54 2011
@@ -1,0 +1,4 @@
+[general]
+
+accept_outofcall_message = yes
+auth_message_requests = no
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Added: asterisk/trunk/tests/channels/SIP/message_unauth/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_unauth/run-test?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_unauth/run-test (added)
+++ asterisk/trunk/tests/channels/SIP/message_unauth/run-test Thu Jun 2 17:20:54 2011
@@ -1,0 +1,44 @@
+#!/usr/bin/env python
+'''
+Copyright (C) 2010, Digium, Inc.
+Russell Bryant <russell at digium.com>
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+'''
+
+import sys
+import os
+
+sys.path.append("lib/python")
+from asterisk.sipp import SIPpTest
+
+
+WORKING_DIR = "/tmp/asterisk-testsuite/sip/message_unauth"
+TEST_DIR = os.path.dirname(os.path.realpath(__file__))
+
+SIPP_SCENARIOS = [
+ {
+ 'scenario' : 'message_recv.xml',
+ '-p' : '5062'
+ },
+ {
+ 'scenario' : 'message.xml',
+ '-p' : '5061'
+ }
+]
+
+
+def main(argv=None):
+ if argv is None:
+ argv = sys.argv
+
+ test = SIPpTest(WORKING_DIR, TEST_DIR, SIPP_SCENARIOS)
+ return test.run()
+
+
+if __name__ == "__main__":
+ sys.exit(main())
+
+
+# vim:sw=4:ts=4:expandtab:textwidth=79
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URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/message_unauth/sipp/message.xml?view=auto&rev=1612
==============================================================================
--- asterisk/trunk/tests/channels/SIP/message_unauth/sipp/message.xml (added)
+++ asterisk/trunk/tests/channels/SIP/message_unauth/sipp/message.xml Thu Jun 2 17:20:54 2011
@@ -1,0 +1,27 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="Basic MESSAGE send and receive">
+ <!-- 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[
+
+ MESSAGE sip:user@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: user <sip:user@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: <sip:user@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 MESSAGE
+ Contact: sip:user@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Expires: 3600
+ Content-Type: text/plain
[... 104 lines stripped ...]
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