[svn-commits] kmoore: testsuite/asterisk/trunk r3462 - in /asterisk/trunk/tests/channels/SI...

[SVN commits to the Digium repositories]

Author: kmoore
Date: Mon Sep 10 08:23:37 2012
New Revision: 3462

URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=3462
Log:
Work around pjsua socket release failure

When pjsua is used with TCP sockets, it fails to release the socket
upon exit instead causing the socket to be in use for up to 60 seconds
after pjsua exits before the socket times out. This change moves the
ports used for TCP sockets outside the range normally used for pjsua
and pjsip instances. This also re-enables the test.

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
    asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/test-config.yaml

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=3462&r1=3461&r2=3462
==============================================================================
--- 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 Mon Sep 10 08:23:37 2012
@@ -7,7 +7,7 @@
 context=transfertest
 type=friend
 host=127.0.0.1
-port=5065
+port=5075
 insecure=invite
 disallow=all
 allow=ulaw
@@ -17,7 +17,7 @@
 context=transfertest
 type=friend
 host=127.0.0.1
-port=5066
+port=5076
 insecure=invite
 disallow=all
 allow=ulaw
@@ -27,7 +27,7 @@
 context=transfertest
 type=friend
 host=127.0.0.1
-port=5067
+port=5077
 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=3462&r1=3461&r2=3462
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/run-test (original)
+++ asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/run-test Mon Sep 10 08:23:37 2012
@@ -31,15 +31,15 @@
         self.final_bridge = 0
 
         # start pjsua clients
-        self.pja = subprocess.Popen(['pjsua', '--local-port=5065',
+        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=5066',
+        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=5067',
+        self.pjc = subprocess.Popen(['pjsua', '--local-port=5077',
                             '--auto-answer=200', '--null-audio', '--no-udp'],
                             stdin=subprocess.PIPE,
                             stdout=subprocess.PIPE)

Modified: asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/test-config.yaml?view=diff&rev=3462&r1=3461&r2=3462
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/test-config.yaml (original)
+++ asterisk/trunk/tests/channels/SIP/sip_attended_transfer_tcp/test-config.yaml Mon Sep 10 08:23:37 2012
@@ -1,5 +1,4 @@
 testinfo:
-    skip: 'Still causing problems.  See ASTERISK-19167'
     summary:     'Test SIP Attended Transfer using TCP endpoints'
     description: |
         'This test verifies the SIP_REFER with Replaces attended transfer routine using TCP endpoints'