[asterisk-commits] jrose: testsuite/asterisk/trunk r4231 - in /asterisk/trunk/tests: cdr/origina...

[SVN commits to the Asterisk project]

Author: jrose
Date: Fri Sep 27 12:05:18 2013
New Revision: 4231

URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=4231
Log:
Testsuite: Include SDPs in 200 OK responses for SIPp scenarios

Addresses some problems that arised with the patch for review 2827

(issue ASTERISK-22424)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2876/

Modified:
    asterisk/trunk/tests/cdr/originate-cdr-disposition/sipp/wait-for-call.xml
    asterisk/trunk/tests/channels/SIP/session_timers/uac_multiple_422_accept/sipp/uas-no-hangup.xml
    asterisk/trunk/tests/channels/SIP/session_timers/uac_multiple_422_originate/sipp/uas-no-hangup.xml

Modified: asterisk/trunk/tests/cdr/originate-cdr-disposition/sipp/wait-for-call.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/cdr/originate-cdr-disposition/sipp/wait-for-call.xml?view=diff&rev=4231&r1=4230&r2=4231
==============================================================================
--- asterisk/trunk/tests/cdr/originate-cdr-disposition/sipp/wait-for-call.xml (original)
+++ asterisk/trunk/tests/cdr/originate-cdr-disposition/sipp/wait-for-call.xml Fri Sep 27 12:05:18 2013
@@ -36,7 +36,19 @@
         [last_Call-ID:]
         [last_CSeq:]
         Contact: sip:[field0 line="[$line]"]@[local_ip]:[local_port]
-        Content-Length: 0
+		Content-Type: application/sdp
+		Content-Length: [len]
+
+		v=0
+		o=- 1324901698 1324901698 IN IP4 [local_ip]
+		s=Polycom IP Phone
+		c=IN IP4 [local_ip]
+		t=0 0
+		a=sendrecv
+		m=audio 2226 RTP/AVP 0 101
+		a=sendrecv
+		a=rtpmap:0 PCMU/8000
+		a=rtpmap:101 telephone-event/8000
 
         ]]>
     </send>

Modified: asterisk/trunk/tests/channels/SIP/session_timers/uac_multiple_422_accept/sipp/uas-no-hangup.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/session_timers/uac_multiple_422_accept/sipp/uas-no-hangup.xml?view=diff&rev=4231&r1=4230&r2=4231
==============================================================================
--- asterisk/trunk/tests/channels/SIP/session_timers/uac_multiple_422_accept/sipp/uas-no-hangup.xml (original)
+++ asterisk/trunk/tests/channels/SIP/session_timers/uac_multiple_422_accept/sipp/uas-no-hangup.xml Fri Sep 27 12:05:18 2013
@@ -135,7 +135,18 @@
       Session-Expires: 256;refresher=uas
       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
       Content-Type: application/sdp
-      Content-Length: 0
+      Content-Length: [len]
+
+      v=0
+      o=- 1324901698 1324901698 IN IP4 [local_ip]
+      s=Polycom IP Phone
+      c=IN IP4 [local_ip]
+      t=0 0
+      a=sendrecv
+      m=audio 2226 RTP/AVP 0 101
+      a=sendrecv
+      a=rtpmap:0 PCMU/8000
+      a=rtpmap:101 telephone-event/8000
 
     ]]>
   </send>

Modified: asterisk/trunk/tests/channels/SIP/session_timers/uac_multiple_422_originate/sipp/uas-no-hangup.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/session_timers/uac_multiple_422_originate/sipp/uas-no-hangup.xml?view=diff&rev=4231&r1=4230&r2=4231
==============================================================================
--- asterisk/trunk/tests/channels/SIP/session_timers/uac_multiple_422_originate/sipp/uas-no-hangup.xml (original)
+++ asterisk/trunk/tests/channels/SIP/session_timers/uac_multiple_422_originate/sipp/uas-no-hangup.xml Fri Sep 27 12:05:18 2013
@@ -150,7 +150,18 @@
       Session-Expires: 256;refresher=uas
       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
       Content-Type: application/sdp
-      Content-Length: 0
+      Content-Length: [len]
+
+      v=0
+      o=- 1324901698 1324901698 IN IP4 [local_ip]
+      s=Polycom IP Phone
+      c=IN IP4 [local_ip]
+      t=0 0
+      a=sendrecv
+      m=audio 2226 RTP/AVP 0 101
+      a=sendrecv
+      a=rtpmap:0 PCMU/8000
+      a=rtpmap:101 telephone-event/8000
 
     ]]>
   </send>