[asterisk-commits] mmichelson: testsuite/asterisk/trunk r3481 - in /asterisk/trunk/tests: channe...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Oct 5 11:25:39 CDT 2012
Author: mmichelson
Date: Fri Oct 5 11:25:34 2012
New Revision: 3481
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=3481
Log:
Rewrite SIP outbound address test.
This gets rid of the old Lua test in favor
of a test that uses the SIPpTestCase. The
test has been simplified to reduce the chance
of bounces.
Review: https://reviewboard.asterisk.org/r/2131
Added:
asterisk/trunk/tests/channels/SIP/sip_outbound_address/originator.py (with props)
asterisk/trunk/tests/channels/SIP/sip_outbound_address/sipp/uas.xml (with props)
Removed:
asterisk/trunk/tests/channels/SIP/sip_outbound_address/run-test
asterisk/trunk/tests/channels/SIP/sip_outbound_address/sipp/dtmf_2833_1.pcap
asterisk/trunk/tests/channels/SIP/sip_outbound_address/sipp/dtmf_2833_2.pcap
asterisk/trunk/tests/channels/SIP/sip_outbound_address/sipp/dtmf_2833_pound.pcap
asterisk/trunk/tests/channels/SIP/sip_outbound_address/sipp/uas1.xml
asterisk/trunk/tests/channels/SIP/sip_outbound_address/sipp/uas2.xml
asterisk/trunk/tests/channels/SIP/sip_outbound_address/test.lua
Modified:
asterisk/trunk/tests/channels/SIP/sip_outbound_address/configs/ast1/extensions.conf
asterisk/trunk/tests/channels/SIP/sip_outbound_address/configs/ast1/sip.conf
asterisk/trunk/tests/channels/SIP/sip_outbound_address/test-config.yaml
asterisk/trunk/tests/manager/login/run-test
Modified: asterisk/trunk/tests/channels/SIP/sip_outbound_address/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_outbound_address/configs/ast1/extensions.conf?view=diff&rev=3481&r1=3480&r2=3481
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_outbound_address/configs/ast1/extensions.conf (original)
+++ asterisk/trunk/tests/channels/SIP/sip_outbound_address/configs/ast1/extensions.conf Fri Oct 5 11:25:34 2012
@@ -1,34 +1,30 @@
-[test_context]
+[default]
;Base line.
;Result: Use sip.conf info
-exten => test1,1,Dial(SIP/peer,,M(readdtmf))
+exten => test1,1,Dial(SIP/bob)
exten => test1,n,Hangup
;Basic change to outbound. Should dial to explicit host:port
;Result: Use host in dialstring
-exten => test2,1,Dial(SIP/peer//127.0.0.1:5062,,M(readdtmf))
+exten => test2,1,Dial(SIP/bob//127.0.0.1:5062)
exten => test2,n,Hangup
;Adding user portion should not affect routing
;Result: Use host in dialstring
-exten => test3,1,Dial(SIP/1000 at peer//127.0.0.1:5062,,M(readdtmf))
+exten => test3,1,Dial(SIP/1000 at bob//127.0.0.1:5062)
exten => test3,n,Hangup
-;Neither should adding extension between peer and host
+;Neither should adding extension between bob and host
;Result: Use host in dialstring
-exten => test4,1,Dial(SIP/peer/1000/127.0.0.1:5062,,M(readdtmf))
+exten => test4,1,Dial(SIP/bob/1000/127.0.0.1:5062)
exten => test4,n,Hangup
;Neither should a conflict in which extension to use
;Result: Use host in dialstring
-exten => test5,1,Dial(SIP/1000 at peer/1000/127.0.0.1:5062,,M(readdtmf))
+exten => test5,1,Dial(SIP/1000 at bob/1000/127.0.0.1:5062)
exten => test5,n,Hangup
;This one has the wrong number of slashes before the host
;Result: Use sip.conf info
-exten => test6,1,Dial(SIP/1000 at peer/127.0.0.1:5062,,M(readdtmf))
+exten => test6,1,Dial(SIP/1000 at bob/127.0.0.1:5062)
exten => test6,n,Hangup
-
-[macro-readdtmf]
-exten => s,1,Read(READRESULT,,1,,1,5)
-exten => s,n,MacroExit
Modified: asterisk/trunk/tests/channels/SIP/sip_outbound_address/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_outbound_address/configs/ast1/sip.conf?view=diff&rev=3481&r1=3480&r2=3481
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_outbound_address/configs/ast1/sip.conf (original)
+++ asterisk/trunk/tests/channels/SIP/sip_outbound_address/configs/ast1/sip.conf Fri Oct 5 11:25:34 2012
@@ -3,8 +3,12 @@
;This is the peer dialed for
;all outbound calls
-[peer]
+[bob]
type=friend
-dtmfmode=rfc2833
host=127.0.0.1
port=5061
+
+[alice]
+type=friend
+host=127.0.0.1
+port=5062
Added: asterisk/trunk/tests/channels/SIP/sip_outbound_address/originator.py
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_outbound_address/originator.py?view=auto&rev=3481
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_outbound_address/originator.py (added)
+++ asterisk/trunk/tests/channels/SIP/sip_outbound_address/originator.py Fri Oct 5 11:25:34 2012
@@ -1,0 +1,57 @@
+#!/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 logging
+
+sys.path.append("lib/python")
+
+LOGGER = logging.getLogger(__name__)
+
+DESTINATIONS = [
+ 'test1',
+ 'test2',
+ 'test3',
+ 'test4',
+ 'test5',
+ 'test6',
+]
+
+class Originator(object):
+ def __init__(self, module_config, test_object):
+ test_object.register_ami_observer(self.ami_connect)
+ test_object.register_scenario_started_observer(self.scenario_started)
+ self.test_object = test_object
+ self.current_destination = 0
+
+ def ami_connect(self, ami):
+ LOGGER.info("AMI connected")
+ self.ami = ami
+
+ def success(self, result):
+ LOGGER.info("Originate Successful")
+ self.current_destination += 1
+
+ def originate_call(self):
+ def failure(result):
+ self.test_object.set_passed(False)
+ return result
+
+ dest = DESTINATIONS[self.current_destinations]
+
+ LOGGER.info("Originating call to %s" % dest)
+
+ deferred = self.ami.originate(channel='Local/%s at default' % dest,
+ application='Echo')
+ deferred.addCallback(self.success).addErrback(failure)
+
+ def scenario_started(self, result):
+ LOGGER.info("Scenario started. Originating call")
+ self.originate_call()
+ return result
Propchange: asterisk/trunk/tests/channels/SIP/sip_outbound_address/originator.py
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/sip_outbound_address/originator.py
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/sip_outbound_address/originator.py
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/sip_outbound_address/sipp/uas.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_outbound_address/sipp/uas.xml?view=auto&rev=3481
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_outbound_address/sipp/uas.xml (added)
+++ asterisk/trunk/tests/channels/SIP/sip_outbound_address/sipp/uas.xml Fri Oct 5 11:25:34 2012
@@ -1,0 +1,106 @@
+<?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 'uas' scenario. -->
+<!-- -->
+
+<scenario name="Basic UAS responder">
+ <Global variables="remote_tag" />
+ <recv request="INVITE" crlf="true">
+ <!-- Save the from tag. We'll need it when we send our BYE -->
+ <action>
+ <ereg regexp=".*(;tag=.*)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="remote_tag"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 180 Ringing
+ [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[
+
+ 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-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 [media_port] RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv request="ACK"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <pause/>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE 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]SIPpTag01[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>[$remote_tag]
+ [last_Call-ID:]
+ CSeq: [cseq] BYE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200">
+ </recv>
+
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
Propchange: asterisk/trunk/tests/channels/SIP/sip_outbound_address/sipp/uas.xml
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svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/sip_outbound_address/sipp/uas.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/sip_outbound_address/sipp/uas.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: asterisk/trunk/tests/channels/SIP/sip_outbound_address/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_outbound_address/test-config.yaml?view=diff&rev=3481&r1=3480&r2=3481
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_outbound_address/test-config.yaml (original)
+++ asterisk/trunk/tests/channels/SIP/sip_outbound_address/test-config.yaml Fri Oct 5 11:25:34 2012
@@ -1,5 +1,4 @@
testinfo:
- skip: 'Skip while failures are debugged'
summary: 'Test explicit outbound host for SIP calls'
description: |
"This tests the ability to specify an explicit host to send
@@ -15,13 +14,41 @@
issues:
- jira : 'ABE-2153'
+test-modules:
+ add-test-to-search-path: 'True'
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+ modules:
+ -
+ typename: 'originator.Originator'
+
+test-object-config:
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': { 'scenario': 'uas.xml', '-p': '5061' } }
+ -
+ scenarios:
+ - { 'key-args': { 'scenario': 'uas.xml', '-p': '5062' } }
+ -
+ scenarios:
+ - { 'key-args': { 'scenario': 'uas.xml', '-p': '5062' } }
+ -
+ scenarios:
+ - { 'key-args': { 'scenario': 'uas.xml', '-p': '5062' } }
+ -
+ scenarios:
+ - { 'key-args': { 'scenario': 'uas.xml', '-p': '5062' } }
+ -
+ scenarios:
+ - { 'key-args': { 'scenario': 'uas.xml', '-p': '5061' } }
+
properties:
minversion: '1.8.0.0'
dependencies:
- - app : 'bash'
- - app : 'asttest'
+ - python: 'starpy'
- sipp :
- version : 'v3.0'
- feature : 'PCAP'
+ version : 'v3.1'
tags:
- SIP
Modified: asterisk/trunk/tests/manager/login/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/manager/login/run-test?view=diff&rev=3481&r1=3480&r2=3481
==============================================================================
--- asterisk/trunk/tests/manager/login/run-test (original)
+++ asterisk/trunk/tests/manager/login/run-test Fri Oct 5 11:25:34 2012
@@ -9,10 +9,13 @@
import sys
from twisted.internet import reactor
+import logging
+import logging.config
sys.path.append("lib/python")
from asterisk.asterisk import Asterisk
from asterisk.TestCase import TestCase
+LOGGER = logging.getLogger(__name__)
class AMILoginTest(TestCase):
def __init__(self):
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