[svn-commits] mmichelson: testsuite/asterisk/trunk r250 - in /asterisk/trunk/tests: ./ func...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Apr 23 15:35:29 CDT 2010
Author: mmichelson
Date: Fri Apr 23 15:35:26 2010
New Revision: 250
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=250
Log:
Add tests for func_srv and explicit setting of remote SIP IP address or host.
Added:
asterisk/trunk/tests/func_srv/
asterisk/trunk/tests/func_srv/configs/
asterisk/trunk/tests/func_srv/configs/extensions.conf (with props)
asterisk/trunk/tests/func_srv/run-test (with props)
asterisk/trunk/tests/func_srv/test-config.yaml (with props)
asterisk/trunk/tests/func_srv/test.lua (with props)
asterisk/trunk/tests/sip_outbound_address/
asterisk/trunk/tests/sip_outbound_address/configs/
asterisk/trunk/tests/sip_outbound_address/configs/extensions.conf (with props)
asterisk/trunk/tests/sip_outbound_address/configs/sip.conf (with props)
asterisk/trunk/tests/sip_outbound_address/run-test (with props)
asterisk/trunk/tests/sip_outbound_address/sipp/
asterisk/trunk/tests/sip_outbound_address/sipp/dtmf_2833_1.pcap (with props)
asterisk/trunk/tests/sip_outbound_address/sipp/dtmf_2833_2.pcap (with props)
asterisk/trunk/tests/sip_outbound_address/sipp/dtmf_2833_pound.pcap (with props)
asterisk/trunk/tests/sip_outbound_address/sipp/uas1.xml (with props)
asterisk/trunk/tests/sip_outbound_address/sipp/uas2.xml (with props)
asterisk/trunk/tests/sip_outbound_address/test-config.yaml (with props)
asterisk/trunk/tests/sip_outbound_address/test.lua (with props)
Modified:
asterisk/trunk/tests/tests.yaml
Added: asterisk/trunk/tests/func_srv/configs/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/func_srv/configs/extensions.conf?view=auto&rev=250
==============================================================================
--- asterisk/trunk/tests/func_srv/configs/extensions.conf (added)
+++ asterisk/trunk/tests/func_srv/configs/extensions.conf Fri Apr 23 15:35:26 2010
@@ -1,0 +1,22 @@
+[test_context]
+
+exten => test,1,Answer
+exten => test,n,NoOp(SRV TEST)
+exten => test,n,SET(ID=${SRVQUERY(_sip._udp.asteriskcheck.com)})
+exten => test,n,NoOp(Got ${SRVRESULT(${ID},getnum)} results)
+exten => test,n,Set(i=1)
+exten => test,n,While($[${i} <= ${SRVRESULT(${ID},getnum)}])
+exten => test,n,UserEvent(SRV,RecordNum: ${i},Host: ${SRVRESULT(${ID},${i},host)},Port: ${SRVRESULT(${ID},${i},port)},Priority: ${SRVRESULT(${ID},${i},priority)},Weight: ${SRVRESULT(${ID},${i},weight)})
+exten => test,n,Set(i=$[${i} + 1])
+exten => test,n,EndWhile
+
+;Test 2 skips using SRVQUERY and instead passes the service directly to SRVRESULT
+exten => test2,1,Answer
+exten => test2,n,NoOp(SRV TEST)
+exten => test2,n,SET(ID=_sip._udp.asteriskcheck.com)
+exten => test2,n,NoOp(Got ${SRVRESULT(${ID},getnum)} results)
+exten => test2,n,Set(i=1)
+exten => test2,n,While($[${i} <= ${SRVRESULT(${ID},getnum)}])
+exten => test2,n,UserEvent(SRV,RecordNum: ${i},Host: ${SRVRESULT(${ID},${i},host)},Port: ${SRVRESULT(${ID},${i},port)},Priority: ${SRVRESULT(${ID},${i},priority)},Weight: ${SRVRESULT(${ID},${i},weight)})
+exten => test2,n,Set(i=$[${i} + 1])
+exten => test2,n,EndWhile
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Added: asterisk/trunk/tests/func_srv/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/func_srv/run-test?view=auto&rev=250
==============================================================================
--- asterisk/trunk/tests/func_srv/run-test (added)
+++ asterisk/trunk/tests/func_srv/run-test Fri Apr 23 15:35:26 2010
@@ -1,0 +1,3 @@
+#!/bin/bash -e
+
+asttest -a / -s tests/func_srv
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Added: asterisk/trunk/tests/func_srv/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/func_srv/test-config.yaml?view=auto&rev=250
==============================================================================
--- asterisk/trunk/tests/func_srv/test-config.yaml (added)
+++ asterisk/trunk/tests/func_srv/test-config.yaml Fri Apr 23 15:35:26 2010
@@ -1,0 +1,10 @@
+testinfo:
+ summary: 'Test func_srv for correctness'
+ description: |
+ 'This test exercises the SRVQUERY and SRVRESULT dialplan functions
+ to be certain that the results obtained are what are expected.'
+
+properties:
+ minversion: '1.8'
+ dependencies:
+ - app: 'asttest'
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Added: asterisk/trunk/tests/func_srv/test.lua
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/func_srv/test.lua?view=auto&rev=250
==============================================================================
--- asterisk/trunk/tests/func_srv/test.lua (added)
+++ asterisk/trunk/tests/func_srv/test.lua Fri Apr 23 15:35:26 2010
@@ -1,0 +1,94 @@
+srv_record = {}
+function srv_record:new(host, port, priority, weight)
+ local s = {
+ host = host or '',
+ port = port or 0,
+ priority = priority or 0,
+ weight = weight or 0
+ }
+ setmetatable(s, self)
+ self.__index = self
+ return s
+end
+
+function srv_record.__eq(rec1, rec2)
+ -- Helpful for debugging
+ print ("Going to compare the following:")
+ print ("Host 1: " .. rec1.host .. " Host 2: " .. rec2.host)
+ print ("Port 1: " .. rec1.port .. " Port 2: " .. rec2.port)
+ print ("priority 1: " .. rec1.priority .. " priority 2: " .. rec2.priority)
+ print ("weight 1: " .. rec1.weight .. " weight 2: " .. rec2.weight)
+ return rec1.host == rec2.host and rec1.port == rec2.port and rec1.priority == rec2.priority and rec1.weight == rec2.weight
+end
+
+srv_records = {
+ srv_record:new("udpserver1.asteriskcheck.com","5060","0","3"),
+ srv_record:new("udpserver2.asteriskcheck.com","5061","1","0"),
+ srv_record:new("udpserver3.asteriskcheck.com","5060","1","0"),
+ srv_record:new("udpserver4.asteriskcheck.com","5060","65535","65535")
+}
+
+function analyze_srv(event)
+ local record_num = tonumber(event["RecordNum"])
+ local record = srv_record:new(event["Host"], event["Port"], event["Priority"], event["Weight"])
+ --Helpful for debugging
+ print ("User event callback")
+ print ("Got record Host: " .. event["Host"] .. " Port: " .. event["Port"] .. " Priority: " .. event["Priority"] .. " Weight: " .. event["Weight"])
+ if record ~= srv_records[record_num] then
+ -- Since records 2 and 3 share the same priority, it is unpredictable
+ -- which order we'll see them arrive
+ if record_num == 2 then
+ record_num = 3
+ elseif record_num == 3 then
+ record_num = 2
+ else
+ fail("Records don't match!")
+ end
+ if record ~= srv_records[record_num] then
+ fail("Records don't match!")
+ end
+ end
+end
+
+function do_call(man, exten)
+ local orig = ast.manager.action:new("Originate")
+ orig["Channel"] = "Local/" .. exten .. "@test_context"
+ orig["Application"] = "Wait"
+ orig["Data"] = "3"
+ man:register_event("UserEvent", analyze_srv)
+ local res, err = man(orig)
+ if not res then
+ fail("Error originating call: " .. err)
+ end
+ if res["Response"] ~= "Success" then
+ fail("Response failure for Originate: " .. res["Message"])
+ end
+ posix.sleep(4)
+ res, err = man:pump_messages()
+ if not res then
+ fail("Error pumping messages: " .. err)
+ end
+ man:process_events()
+ man:unregister_event("UserEvent", analyze_srv)
+end
+
+instance = ast.new()
+instance:load_config("configs/extensions.conf")
+instance:generate_manager_conf()
+instance:spawn()
+
+man,err = instance:manager_connect()
+if not man then
+ fail("error connecting to asterisk: " .. err)
+end
+
+login = ast.manager.action.login()
+if not login then
+ fail("Couldn't create login?")
+end
+man(login)
+do_call(man, "test")
+do_call(man, "test2")
+logoff = ast.manager.action.logoff()
+man(logoff)
+instance:term_or_kill()
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Added: asterisk/trunk/tests/sip_outbound_address/configs/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/sip_outbound_address/configs/extensions.conf?view=auto&rev=250
==============================================================================
--- asterisk/trunk/tests/sip_outbound_address/configs/extensions.conf (added)
+++ asterisk/trunk/tests/sip_outbound_address/configs/extensions.conf Fri Apr 23 15:35:26 2010
@@ -1,0 +1,35 @@
+[test_context]
+
+;Base line.
+;Result: Use sip.conf info
+exten => test1,1,Dial(SIP/peer,,M(readdtmf))
+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,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,n,Hangup
+
+;Neither should adding extension between peer and host
+;Result: Use host in dialstring
+exten => test4,1,Dial(SIP/peer/1000/127.0.0.1:5062,,M(readdtmf))
+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,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,n,Hangup
+
+[macro-readdtmf]
+exten => s,1,Read(READRESULT,,1,,1,5)
+exten => s,n,MacroExit
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Added: asterisk/trunk/tests/sip_outbound_address/configs/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/sip_outbound_address/configs/sip.conf?view=auto&rev=250
==============================================================================
--- asterisk/trunk/tests/sip_outbound_address/configs/sip.conf (added)
+++ asterisk/trunk/tests/sip_outbound_address/configs/sip.conf Fri Apr 23 15:35:26 2010
@@ -1,0 +1,10 @@
+[general]
+canreinvite=no
+
+;This is the peer dialed for
+;all outbound calls
+[peer]
+type=friend
+dtmfmode=rfc2833
+host=127.0.0.1
+port=5061
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Added: asterisk/trunk/tests/sip_outbound_address/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/sip_outbound_address/run-test?view=auto&rev=250
==============================================================================
--- asterisk/trunk/tests/sip_outbound_address/run-test (added)
+++ asterisk/trunk/tests/sip_outbound_address/run-test Fri Apr 23 15:35:26 2010
@@ -1,0 +1,3 @@
+#!/bin/bash -e
+
+asttest -a / -s tests/sip_outbound_address
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Added: asterisk/trunk/tests/sip_outbound_address/sipp/dtmf_2833_1.pcap
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/sip_outbound_address/sipp/dtmf_2833_1.pcap?view=auto&rev=250
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Added: asterisk/trunk/tests/sip_outbound_address/sipp/dtmf_2833_2.pcap
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Added: asterisk/trunk/tests/sip_outbound_address/sipp/dtmf_2833_pound.pcap
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/sip_outbound_address/sipp/dtmf_2833_pound.pcap?view=auto&rev=250
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Added: asterisk/trunk/tests/sip_outbound_address/sipp/uas1.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/sip_outbound_address/sipp/uas1.xml?view=auto&rev=250
==============================================================================
--- asterisk/trunk/tests/sip_outbound_address/sipp/uas1.xml (added)
+++ asterisk/trunk/tests/sip_outbound_address/sipp/uas1.xml Fri Apr 23 15:35:26 2010
@@ -1,0 +1,133 @@
+<?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">
+ <!-- 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 request="INVITE" crlf="true">
+ </recv>
+
+ <!-- The '[last_*]' keyword is replaced automatically by the -->
+ <!-- specified header if it was present in the last message received -->
+ <!-- (except if it was a retransmission). If the header was not -->
+ <!-- present or if no message has been received, the '[last_*]' -->
+ <!-- keyword is discarded, and all bytes until the end of the line -->
+ <!-- are also discarded. -->
+ <!-- -->
+ <!-- If the specified header was present several times in the -->
+ <!-- message, all occurences are concatenated (CRLF seperated) -->
+ <!-- to be used in place of the '[last_*]' keyword. -->
+
+ <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 0 101
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ a=fmtp:101 0-16
+ a=sendrecv
+
+ ]]>
+ </send>
+
+ <recv request="ACK"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <nop>
+ <action>
+ <exec play_pcap_audio="sipp/dtmf_2833_1.pcap" />
+ </action>
+ </nop>
+
+ <pause milliseconds="200" />
+
+ <nop>
+ <action>
+ <exec play_pcap_audio="sipp/dtmf_2833_pound.pcap" />
+ </action>
+ </nop>
+
+ <pause milliseconds="200" />
+
+ <recv request="BYE">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- Keep the call open for a while in case the 200 is lost to be -->
+ <!-- able to retransmit it if we receive the BYE again. -->
+ <pause milliseconds="4000"/>
+
+
+ <!-- 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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==============================================================================
--- asterisk/trunk/tests/sip_outbound_address/sipp/uas2.xml (added)
+++ asterisk/trunk/tests/sip_outbound_address/sipp/uas2.xml Fri Apr 23 15:35:26 2010
@@ -1,0 +1,133 @@
+<?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">
+ <!-- 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 request="INVITE" crlf="true">
+ </recv>
+
+ <!-- The '[last_*]' keyword is replaced automatically by the -->
+ <!-- specified header if it was present in the last message received -->
+ <!-- (except if it was a retransmission). If the header was not -->
+ <!-- present or if no message has been received, the '[last_*]' -->
+ <!-- keyword is discarded, and all bytes until the end of the line -->
+ <!-- are also discarded. -->
+ <!-- -->
+ <!-- If the specified header was present several times in the -->
+ <!-- message, all occurences are concatenated (CRLF seperated) -->
+ <!-- to be used in place of the '[last_*]' keyword. -->
+
+ <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 0 101
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ a=fmtp:101 0-16
+ a=sendrecv
+
+ ]]>
+ </send>
+
+ <recv request="ACK"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <nop>
+ <action>
+ <exec play_pcap_audio="sipp/dtmf_2833_2.pcap" />
+ </action>
+ </nop>
+
+ <pause milliseconds="200" />
+
+ <nop>
+ <action>
+ <exec play_pcap_audio="sipp/dtmf_2833_pound.pcap" />
+ </action>
+ </nop>
+
+ <pause milliseconds="200" />
+
+ <recv request="BYE">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- Keep the call open for a while in case the 200 is lost to be -->
+ <!-- able to retransmit it if we receive the BYE again. -->
+ <pause milliseconds="4000"/>
+
+
+ <!-- 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/trunk/tests/sip_outbound_address/sipp/uas2.xml
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svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/sip_outbound_address/sipp/uas2.xml
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svn:mime-type = text/plain
Added: asterisk/trunk/tests/sip_outbound_address/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/sip_outbound_address/test-config.yaml?view=auto&rev=250
==============================================================================
--- asterisk/trunk/tests/sip_outbound_address/test-config.yaml (added)
+++ asterisk/trunk/tests/sip_outbound_address/test-config.yaml Fri Apr 23 15:35:26 2010
@@ -1,0 +1,22 @@
+testinfo:
+ summary: 'Test explicit outbound host for SIP calls'
+ description: |
+ "This tests the ability to specify an explicit host to send
+ an outbound SIP INVITE to. The test works by having two SIPp
+ UASs waiting for INVITEs. Each is set to answer an inbound INVITE
+ and then send an RFC2833 DTMF digit. Asterisk will read this digit,
+ and determine who answered the call. Based on who answered the call,
+ we can determine which UAS answered and whether it was the one we
+ expected.
+
+ It is possible that this test could be done in a more straightforward
+ manner. Suggestions are certainly welcome."
+ issues:
+ - jira : 'ABE-2153'
+
+properties:
+ minversion: 'trunk'
+ dependencies:
+ - app : 'bash'
+ - app : 'sipp'
+ - app : 'asttest'
Propchange: asterisk/trunk/tests/sip_outbound_address/test-config.yaml
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svn:eol-style = native
Propchange: asterisk/trunk/tests/sip_outbound_address/test-config.yaml
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svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/sip_outbound_address/test-config.yaml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/sip_outbound_address/test.lua
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/sip_outbound_address/test.lua?view=auto&rev=250
==============================================================================
--- asterisk/trunk/tests/sip_outbound_address/test.lua (added)
+++ asterisk/trunk/tests/sip_outbound_address/test.lua Fri Apr 23 15:35:26 2010
@@ -1,0 +1,123 @@
+function sipp_exec(scenario, local_port)
+ return proc.exec_io("sipp",
+ "127.0.0.1",
+ "-m", "1",
+ "-sf", scenario,
+ "-i", "127.0.0.1",
+ "-p", local_port,
+ "-timeout", "30",
+ "-trace_err"
+ )
+end
+
+function sipp_exec_and_wait(scenario, name, local_port)
+ return sipp_check_error(sipp_exec(scenario, name, local_port), scenario)
+end
+
+function sipp_check_error(p, scenario)
+ local res, err = p:wait()
+
+ if not res then error(err) end
+ if res ~= 0 then
+ error("error while executing " .. scenario .. " sipp scenario (sipp exited with status " .. res .. ")\n" .. p.stderr:read("*a"))
+ end
+
+ return res, err
+end
+
+function varset_event_one(event)
+ if (event["Variable"] == "READRESULT") then
+ if (event["Value"] ~= "1") then
+ fail("Wrong UAS answered. Expected DTMF '1' but received " .. event["Value"])
+ else
+ passed = true;
+ end
+ end
+end
+
+function varset_event_two(event)
+ if (event["Variable"] == "READRESULT") then
+ if (event["Value"] ~= "2") then
+ fail("Wrong UAS answered. Expected DTMF '2' but received " .. event["Value"])
+ else
+ passed = true;
+ end
+ end
+end
+
+function manager_setup(a)
+ m,err = a:manager_connect()
+ if not m then
+ fail("error connecting to asterisk: " .. err)
+ end
+
+ login = ast.manager.action:new("login")
+ login["Username"] = "asttest"
+ login["Secret"] = "asttest"
+
+ local r = m(login)
+ if not r then
+ fail("error logging in to the manager: " .. err)
+ end
+
+ if r["Response"] ~= "Success" then
+ fail("error authenticating: " .. r["Message"])
+ end
+end
+
+function setup_uas()
+ local t1 = sipp_exec("sipp/uas1.xml", "5061")
+ local t2 = sipp_exec("sipp/uas2.xml", "5062")
+ return t1, t2
+end
+
+function kill_uas(t1, t2)
+ t1:term_or_kill()
+ t2:term_or_kill()
+end
+
+function test_call(exten, handler)
+ passed = false;
+ m:register_event("VarSet", handler)
+ local uas1, uas2 = setup_uas()
+ local orig = ast.manager.action:new("Originate")
+ orig["Channel"] = "Local/" .. exten .. "@test_context"
+ orig["Context"] = "test_context"
+ orig["Application"] = "Wait"
+ orig["Data"] = "1"
+ local res, err = m(orig)
+ if not res then
+ fail("Error originating call: " .. err)
+ end
+ if res["Response"] ~= "Success" then
+ fail("Response failure for Originate: " .. res["Message"])
+ end
+ posix.sleep(2)
+ local res, err = m:pump_messages()
+ if not res then
+ fail("error pumping manager messages: " .. err)
+ end
+ m:process_events()
+ m:unregister_event("VarSet", handler)
+ kill_uas(uas1, uas2)
+ if not passed then
+ fail("Failure has occurred")
+ end
+end
+
+a = ast.new()
+a:load_config("configs/sip.conf")
+a:load_config("configs/extensions.conf")
+a:generate_manager_conf()
+a:spawn()
+
+manager_setup(a)
+
+test_call("test1", varset_event_one)
+test_call("test2", varset_event_two)
+test_call("test3", varset_event_two)
+test_call("test4", varset_event_two)
+test_call("test5", varset_event_two)
+test_call("test6", varset_event_one)
+
+a:term_or_kill()
Propchange: asterisk/trunk/tests/sip_outbound_address/test.lua
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svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/sip_outbound_address/test.lua
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svn:mime-type = text/plain
Modified: asterisk/trunk/tests/tests.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/tests.yaml?view=diff&rev=250&r1=249&r2=250
==============================================================================
--- asterisk/trunk/tests/tests.yaml (original)
+++ asterisk/trunk/tests/tests.yaml Fri Apr 23 15:35:26 2010
@@ -8,3 +8,5 @@
- test: 'iax-call-basic'
- test: 'manager-action-events-response'
- test: 'originate-cdr-disposition'
+ - test: 'func_srv'
+ - test: 'sip_outbound_address'
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