[svn-commits] twilson: testsuite/asterisk/trunk r2877 - in /asterisk/trunk/tests/channels/S...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Dec 12 10:51:43 CST 2011
Author: twilson
Date: Mon Dec 12 10:51:40 2011
New Revision: 2877
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=2877
Log:
Fix and re-enable SRTP tests
Modified:
asterisk/trunk/tests/channels/SIP/noload_res_srtp/run-test
asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/sip.conf
asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/run-test
asterisk/trunk/tests/channels/SIP/sip_srtp/run-test
asterisk/trunk/tests/channels/SIP/tests.yaml
Modified: asterisk/trunk/tests/channels/SIP/noload_res_srtp/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/noload_res_srtp/run-test?view=diff&rev=2877&r1=2876&r2=2877
==============================================================================
--- asterisk/trunk/tests/channels/SIP/noload_res_srtp/run-test (original)
+++ asterisk/trunk/tests/channels/SIP/noload_res_srtp/run-test Mon Dec 12 10:51:40 2011
@@ -56,6 +56,10 @@
else:
print "Don't know which side is connected."
+ if self.connected_chan1 and self.connected_no_srtp1 and self.connected_chan2 and self.connected_no_srtp2:
+ print "Test passed"
+ self.passed = True
+ reactor.stop()
# Hold the AGI connection until the reactor times out
# so the other side has a chance to get its test result.
## Drop the AGI connection
@@ -71,9 +75,6 @@
print "self.connected_no_srtp1:%s" % (self.connected_no_srtp1)
print "self.connected_chan2: %s" % (self.connected_chan2)
print "self.connected_no_srtp2:%s" % (self.connected_no_srtp2)
- if self.connected_chan1 and self.connected_no_srtp1 and self.connected_chan2 and self.connected_no_srtp2:
- print "Test passed"
- self.passed = True
def main():
test = SIPCallTest()
Modified: asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/sip.conf?view=diff&rev=2877&r1=2876&r2=2877
==============================================================================
--- asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/sip.conf (original)
+++ asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/configs/ast1/sip.conf Mon Dec 12 10:51:40 2011
@@ -6,6 +6,7 @@
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
sipdebug=yes
+storesipcause=yes
[authentication]
Modified: asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/run-test?view=diff&rev=2877&r1=2876&r2=2877
==============================================================================
--- asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/run-test (original)
+++ asterisk/trunk/tests/channels/SIP/noload_res_srtp_attempt_srtp/run-test Mon Dec 12 10:51:40 2011
@@ -48,8 +48,7 @@
self.ami[ami.id].registerEvent("VarSet", self.ami_test_varset)
print "Initiating test call"
- self.ast[0].cli_exec(
- "originate SIP/2000/2000 extension 1000 at siptest1")
+ self.ast[0].cli_originate("SIP/2000/2000 extension 1000 at siptest1", blocking=False)
# This is called whenever an AMI VarSet event occurs on Ast1.
def ami_test_varset(self, ami, event):
@@ -61,7 +60,7 @@
return
print " Value of event[variable] = %s" % (event["variable"])
print " Value of event[value] = %s" % (event["value"])
- if 0 <= event["variable"].startswith("~HASH~SIP_CAUSE~"):
+ if event["variable"].startswith("~HASH~SIP_CAUSE~"):
cause_code = event["value"].split(" ")[1]
if cause_code == "488":
# The call failed for the expected reason "488 Not acceptable".
Modified: asterisk/trunk/tests/channels/SIP/sip_srtp/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_srtp/run-test?view=diff&rev=2877&r1=2876&r2=2877
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_srtp/run-test (original)
+++ asterisk/trunk/tests/channels/SIP/sip_srtp/run-test Mon Dec 12 10:51:40 2011
@@ -36,8 +36,7 @@
TestCase.run(self)
print "Initiating test call"
- self.ast[0].cli_exec(
- "originate SIP/2000/2000 extension 1000 at siptest1")
+ self.ast[0].cli_originate("SIP/2000/2000 extension 1000 at siptest1")
# This is called by each Asterisk instance if the call gets connected.
def fastagi_connect(self, agi):
@@ -56,6 +55,10 @@
else:
print "Don't know which side is connected."
+ if self.connected_chan1 and self.connected_srtp1 and self.connected_chan2 and self.connected_srtp2:
+ print "Test passed"
+ self.passed = True
+ reactor.stop()
# Hold the AGI connection until the reactor times out
# so the other side has a chance to get its test result.
## Drop the AGI connection
@@ -65,15 +68,6 @@
def stop_asterisk(self):
TestCase.stop_asterisk(self)
-
- # Determine if the test passed
- print "self.connected_chan1:%s" % (self.connected_chan1)
- print "self.connected_srtp1:%s" % (self.connected_srtp1)
- print "self.connected_chan2:%s" % (self.connected_chan2)
- print "self.connected_srtp2:%s" % (self.connected_srtp2)
- if self.connected_chan1 and self.connected_srtp1 and self.connected_chan2 and self.connected_srtp2:
- print "Test passed"
- self.passed = True
def main():
test = SIPCallTest()
Modified: asterisk/trunk/tests/channels/SIP/tests.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/tests.yaml?view=diff&rev=2877&r1=2876&r2=2877
==============================================================================
--- asterisk/trunk/tests/channels/SIP/tests.yaml (original)
+++ asterisk/trunk/tests/channels/SIP/tests.yaml Mon Dec 12 10:51:40 2011
@@ -17,10 +17,10 @@
- test: 'sip_channel_params'
- test: 'sip_tls_call'
- test: 'sip_tls_register'
-# - test: 'sip_srtp'
+ - test: 'sip_srtp'
- test: 'noload_res_srtp'
-# - test: 'noload_res_srtp_attempt_srtp'
-# - test: 'secure_bridge_media'
+ - test: 'noload_res_srtp_attempt_srtp'
+ - test: 'secure_bridge_media'
- test: 'message_disabled'
- test: 'message_unauth'
- test: 'message_auth'
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