[asterisk-commits] mmichelson: testsuite/asterisk/trunk r402 - in /asterisk/trunk/tests/queues/r...

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Tue Jun 15 09:57:12 2010
New Revision: 402

URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=402
Log:
Fix codec answer in sipp scenario.

Honestly, I have no idea how this was ever passing before.


Modified:
    asterisk/trunk/tests/queues/ringinuse_and_pause/configs/sip.conf
    asterisk/trunk/tests/queues/ringinuse_and_pause/sipp/uas.xml

Modified: asterisk/trunk/tests/queues/ringinuse_and_pause/configs/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/queues/ringinuse_and_pause/configs/sip.conf?view=diff&rev=402&r1=401&r2=402
==============================================================================
--- asterisk/trunk/tests/queues/ringinuse_and_pause/configs/sip.conf (original)
+++ asterisk/trunk/tests/queues/ringinuse_and_pause/configs/sip.conf Tue Jun 15 09:57:12 2010
@@ -1,4 +1,5 @@
 [general]
+sipdebug=yes
 udpbindaddr=127.0.0.1:5060
 canreinvite=no
 videosupport=yes

Modified: asterisk/trunk/tests/queues/ringinuse_and_pause/sipp/uas.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/queues/ringinuse_and_pause/sipp/uas.xml?view=diff&rev=402&r1=401&r2=402
==============================================================================
--- asterisk/trunk/tests/queues/ringinuse_and_pause/sipp/uas.xml (original)
+++ asterisk/trunk/tests/queues/ringinuse_and_pause/sipp/uas.xml Tue Jun 15 09:57:12 2010
@@ -71,8 +71,8 @@
       c=IN IP[media_ip_type] [media_ip]
       t=0 0
       m=audio [media_port] RTP/AVP 0
-      a=rtpmap:0 ulaw/8000
-	  a=sendrecv
+      a=rtpmap:0 PCMU/8000
+      a=sendrecv
 
     ]]>
   </send>