[asterisk-commits] mmichelson: branch mmichelson/phone-testsuite r3141 - in /asterisk/team/mmich...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Mar 23 16:30:10 CDT 2012
Author: mmichelson
Date: Fri Mar 23 16:30:06 2012
New Revision: 3141
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=3141
Log:
Progress towards a SIP custom presence test.
No where near complete. The scenarios are probably error-prone.
Will get around to finishing Monday.
Added:
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/run-test (with props)
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen1.txt (with props)
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen2.txt (with props)
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen3.txt (with props)
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen4.txt (with props)
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/test-config.yaml (with props)
Modified:
asterisk/team/mmichelson/phone-testsuite/tests/func_presencestate/run-test
Added: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/run-test?view=auto&rev=3141
==============================================================================
--- asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/run-test (added)
+++ asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/run-test Fri Mar 23 16:30:06 2012
@@ -1,0 +1,50 @@
+#!/usr/bin/env python
+'''
+Copyright (C) 2012, Digium, Inc.
+Mark Michelson <mmichelson at digium.com>
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+'''
+import sys
+import os
+import logging
+
+sys.path.append("lib/python")
+
+from asterisk.asterisk import Asterisk
+from asterisk.TestCase import TestCase
+from asterisk.sipp import SIPpScenario
+from twisted.internet import reactor
+
+logger = logging.getLogger(__name__)
+TEST_DIR = os.path.dirname(os.path.realpath(__file__))
+
+SIPP_SCENARIOS = [
+ {'scenario' : 'scen1.xml',},
+ {'scenario' : 'scen2.xml',},
+ {'scenario' : 'scen3.xml',},
+ {'scenario' : 'scen4.xml',},
+]
+
+class SIPCustomPresence(TestCase):
+ def __init__(self):
+ TestCase.__init__(self)
+ self.create_asterisk
+
+ def ami_connect(self, ami):
+
+ def run(self):
+ TestCase.run(self)
+ self.create_ami_factory()
+
+def main()
+ test = SIPCustomPresence()
+ reactor.run()
+ if test.passed:
+ return 0
+ else:
+ return 1
+
+if __name__ == "__main__":
+ sys.exit(main())
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/run-test
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/run-test
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/run-test
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen1.txt
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen1.txt?view=auto&rev=3141
==============================================================================
--- asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen1.txt (added)
+++ asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen1.txt Fri Mar 23 16:30:06 2012
@@ -1,0 +1,126 @@
+<?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 default 'uac' scenario. -->
+<!-- -->
+
+<scenario name="Basic Sipstone UAC">
+ <!-- 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[
+
+ SUBSCRIBE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 SUBSCRIBE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Event: presence
+ User-Agent: A Digium Phone
+ Accept: application/pidf+xml
+ Expires: 3600
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <!-- 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">
+ </recv>
+
+ <!-- This delay can be customized by the -d command-line option -->
+ <!-- or by adding a 'milliseconds = "value"' option here. -->
+ <pause/>
+
+ <!-- Initial NOTIFY upon subscribing -->
+ <recv request="NOTIFY" crlf="true" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- NOTIFY indicating presence change -->
+ <recv request="NOTIFY" crlf="true" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ SUBSCRIBE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: [cseq+1] SUBSCRIBE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Event: presence
+ User-Agent: A Digium Phone
+ Accept: application/pidf+xml
+ Expires: 0
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true" />
+
+ <!-- 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>
+
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen1.txt
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen1.txt
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen1.txt
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen2.txt
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen2.txt?view=auto&rev=3141
==============================================================================
--- asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen2.txt (added)
+++ asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen2.txt Fri Mar 23 16:30:06 2012
@@ -1,0 +1,144 @@
+<?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 default 'uac' scenario. -->
+<!-- -->
+
+<scenario name="Basic Sipstone UAC">
+ <!-- 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[
+
+ SUBSCRIBE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 SUBSCRIBE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Event: presence
+ User-Agent: A Digium Phone
+ Accept: application/pidf+xml
+ Expires: 3600
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <!-- 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">
+ </recv>
+
+ <!-- This delay can be customized by the -d command-line option -->
+ <!-- or by adding a 'milliseconds = "value"' option here. -->
+ <pause/>
+
+ <!-- Initial NOTIFY upon subscribing -->
+ <recv request="NOTIFY" crlf="true" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- NOTIFY indicating presence change -->
+ <recv request="NOTIFY" crlf="true" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- NOTIFY indicating second presence change -->
+ <recv request="NOTIFY" crlf="true" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ SUBSCRIBE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: [cseq+1] SUBSCRIBE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Event: presence
+ User-Agent: A Digium Phone
+ Accept: application/pidf+xml
+ Expires: 0
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true" />
+
+ <!-- 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>
+
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen2.txt
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen2.txt
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen2.txt
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen3.txt
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen3.txt?view=auto&rev=3141
==============================================================================
--- asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen3.txt (added)
+++ asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen3.txt Fri Mar 23 16:30:06 2012
@@ -1,0 +1,200 @@
+<?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 default 'uac' scenario. -->
+<!-- -->
+
+<scenario name="Basic Sipstone UAC">
+ <!-- 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[
+
+ SUBSCRIBE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 SUBSCRIBE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Event: presence
+ User-Agent: A Digium Phone
+ Accept: application/pidf+xml
+ Expires: 3600
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <!-- 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">
+ </recv>
+
+ <!-- This delay can be customized by the -d command-line option -->
+ <!-- or by adding a 'milliseconds = "value"' option here. -->
+ <pause/>
+
+ <!-- Initial NOTIFY upon subscribing -->
+ <recv request="NOTIFY" crlf="true" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- NOTIFY indicating presence change -->
+ <recv request="NOTIFY" crlf="true" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ SUBSCRIBE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: [cseq+1] SUBSCRIBE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Event: presence
+ User-Agent: A Digium Phone
+ Accept: application/pidf+xml
+ Expires: 0
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true" />
+
+ <!-- Resubscribe -->
+ <send retrans="500">
+ <![CDATA[
+
+ SUBSCRIBE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 SUBSCRIBE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Event: presence
+ User-Agent: A Digium Phone
+ Accept: application/pidf+xml
+ Expires: 3600
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <!-- 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">
+ </recv>
+
+ <!-- Initial NOTIFY upon subscribing -->
+ <recv request="NOTIFY" crlf="true" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- Now unsubscribe again -->
+ <send retrans="500">
+ <![CDATA[
+
+ SUBSCRIBE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: [cseq+1] SUBSCRIBE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Event: presence
+ User-Agent: A Digium Phone
+ Accept: application/pidf+xml
+ Expires: 0
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true" />
+
+ <!-- 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>
+
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen3.txt
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen3.txt
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen3.txt
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen4.txt
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen4.txt?view=auto&rev=3141
==============================================================================
--- asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen4.txt (added)
+++ asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen4.txt Fri Mar 23 16:30:06 2012
@@ -1,0 +1,103 @@
+<?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 default 'uac' scenario. -->
+<!-- -->
+
+<scenario name="Basic Sipstone UAC">
+ <!-- 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[
+
+ SUBSCRIBE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 SUBSCRIBE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Event: presence
+ User-Agent: A Digium Phone
+ Accept: application/pidf+xml
+ Expires: 3600
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <!-- 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">
+ </recv>
+
+ <!-- This delay can be customized by the -d command-line option -->
+ <!-- or by adding a 'milliseconds = "value"' option here. -->
+ <pause/>
+
+ <!-- Initial NOTIFY upon subscribing -->
+ <recv request="NOTIFY" crlf="true" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- NOTIFY indicating end of subscription -->
+ <recv request="NOTIFY" crlf="true" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- 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>
+
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen4.txt
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen4.txt
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/sipp/scen4.txt
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/test-config.yaml?view=auto&rev=3141
==============================================================================
--- asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/test-config.yaml (added)
+++ asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/test-config.yaml Fri Mar 23 16:30:06 2012
@@ -1,0 +1,11 @@
+testinfo:
+ summary: 'Test Reception of Digium custom presence'
+ description: |
+ 'Subscribe to presence and get notified of presence changes'
+
+properties:
+ minversion: '11'
+ dependencies:
+ - app : 'sipp'
+ tags:
+ - SIP
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/test-config.yaml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/test-config.yaml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/sip_custom_presence/test-config.yaml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: asterisk/team/mmichelson/phone-testsuite/tests/func_presencestate/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/phone-testsuite/tests/func_presencestate/run-test?view=diff&rev=3141&r1=3140&r2=3141
==============================================================================
--- asterisk/team/mmichelson/phone-testsuite/tests/func_presencestate/run-test (original)
+++ asterisk/team/mmichelson/phone-testsuite/tests/func_presencestate/run-test Fri Mar 23 16:30:06 2012
@@ -25,7 +25,7 @@
"base64write"]
self.passes = []
self.passed = False
-
+
def passOrFail(self, ami, event):
if not event.get("userevent") == "Presence":
return
More information about the asterisk-commits
mailing list