[svn-commits] kmoore: testsuite/asterisk/trunk r3469 - in /asterisk/trunk/tests/channels/SI...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Sep 18 09:57:43 CDT 2012
Author: kmoore
Date: Tue Sep 18 09:57:37 2012
New Revision: 3469
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=3469
Log:
Remove dependence on pjsua in broken situations
Previously, the PJSUA binary receiving a call would cause it to fail to
release its socket properly. This replaces the failing instances of
PJSUA with SIPp instances that handle their sockets correctly. PJSUA is
still used for the scenario in which it was working properly.
(closes issue SWP-4660)
Added:
asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/sipp/
asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/sipp/uas-no-hangup.xml (with props)
Modified:
asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/configs/ast1/sip.conf
asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/run-test
Modified: asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/configs/ast1/sip.conf?view=diff&rev=3469&r1=3468&r2=3469
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/configs/ast1/sip.conf (original)
+++ asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/configs/ast1/sip.conf Tue Sep 18 09:57:37 2012
@@ -6,18 +6,14 @@
[end_a]
context=transfertest
type=friend
-host=127.0.0.1
-port=5075
insecure=invite
-disallow=all
-allow=ulaw
transport=tcp
[end_b]
context=transfertest
type=friend
host=127.0.0.1
-port=5076
+port=5066
insecure=invite
disallow=all
allow=ulaw
@@ -27,7 +23,7 @@
context=transfertest
type=friend
host=127.0.0.1
-port=5077
+port=5067
insecure=invite
disallow=all
allow=ulaw
Modified: asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/run-test?view=diff&rev=3469&r1=3468&r2=3469
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/run-test (original)
+++ asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/run-test Tue Sep 18 09:57:37 2012
@@ -15,9 +15,11 @@
sys.path.append("lib/python")
from asterisk.TestCase import TestCase
+from asterisk.sipp import SIPpScenario
import logging
LOGGER = logging.getLogger(__name__)
+TEST_DIR = os.path.dirname(os.path.realpath(__file__))
class AttTransferTestTcp(TestCase):
'''This class handles the testing of attended transfers when using
@@ -27,25 +29,22 @@
TestCase.__init__(self)
self.reactor_timeout = 20
self.create_asterisk()
- self.chans = []
+ self.chans = []
self.final_bridge = 0
- # start pjsua clients
- self.pja = subprocess.Popen(['pjsua', '--local-port=5075',
- '--auto-answer=200', '--null-audio', '--no-udp'],
- stdin=subprocess.PIPE,
- stdout=subprocess.PIPE)
- self.pjb = subprocess.Popen(['pjsua', '--local-port=5076',
- '--auto-answer=200', '--null-audio', '--no-udp'],
- stdin=subprocess.PIPE,
- stdout=subprocess.PIPE)
- self.pjc = subprocess.Popen(['pjsua', '--local-port=5077',
+ # start pjsua client
+ self.pja = subprocess.Popen(['pjsua', '--local-port=5065',
'--auto-answer=200', '--null-audio', '--no-udp'],
stdin=subprocess.PIPE,
stdout=subprocess.PIPE)
def run(self):
'''Handle completion of Asterisk startup.'''
+ # start up sipp scenarios
+ self.sippb = SIPpScenario(TEST_DIR, {'scenario': 'uas-no-hangup.xml', '-t': 't1', '-p': '5066'})
+ self.sippc = SIPpScenario(TEST_DIR, {'scenario': 'uas-no-hangup.xml', '-t': 't1', '-p': '5067'})
+ self.sippb.run(self)
+ self.sippc.run(self)
TestCase.run(self)
self.create_ami_factory()
@@ -56,18 +55,13 @@
self.ami[0].registerEvent('Bridge', self.bridge_callback)
self.ami[0].registerEvent('VarSet', self.bridgepeer_callback)
- # allow pjsua time to setup so that it can actually accept calls
- reactor.callLater(5, self.callAToB)
-
- def callAToB(self):
- '''Kick off the A to B call.'''
LOGGER.info("Kicking off A-to-B call")
self.pja.stdin.write("m\n")
self.pja.stdin.write("sip:call_b at 127.0.0.1:5060;transport=TCP\n")
def bridge_callback(self, ami, event):
'''Handle Bridge events to catch.'''
- self.chans.append(event['channel2'])
+ self.chans.append(event['channel2'])
numchans = len(self.chans)
if numchans == 1:
LOGGER.info("Kicking off A-to-C call")
@@ -98,13 +92,11 @@
def doCleanup(self):
'''Do final cleanup, output debugging, kill off pjsua processes.'''
- self.ami[0].hangup(self.chans[1])
+ self.ami[0].hangup(self.chans[1])
self.ast[0].cli_exec("core show locks") # get lock output in case of deadlock before tearing down.
self.ast[0].cli_exec("core show channels")# if channels are still up for some reason, we want to know that as well
os.kill(self.pja.pid, signal.SIGKILL)
- os.kill(self.pjb.pid, signal.SIGKILL)
- os.kill(self.pjc.pid, signal.SIGKILL)
- self.stop_reactor()
+ self.stop_reactor()
def main():
'''Run SIP TCP attended transfer test.'''
Added: asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/sipp/uas-no-hangup.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/sipp/uas-no-hangup.xml?view=auto&rev=3469
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/sipp/uas-no-hangup.xml (added)
+++ asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/sipp/uas-no-hangup.xml Tue Sep 18 09:57:37 2012
@@ -1,0 +1,78 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic UAS responder">
+ <recv request="INVITE" crlf="true">
+ </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"
+ optional="true"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <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>
+
+ <timewait milliseconds="4000"/>
+
+ <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_attended_transfer_tcp/sipp/uas-no-hangup.xml
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Propchange: asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/sipp/uas-no-hangup.xml
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svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/sipp/uas-no-hangup.xml
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