[asterisk-commits] jrose: testsuite/asterisk/trunk r4117 - in /asterisk/trunk/tests/channels/SIP...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Sep 5 14:23:34 CDT 2013
Author: jrose
Date: Thu Sep 5 14:23:32 2013
New Revision: 4117
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=4117
Log:
Add SDP to 200 OKs for the non-directmedia SIP Hold test sipp scenarios
Modified:
asterisk/trunk/tests/channels/SIP/alwaysauthreject/test-config.yaml
asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A.xml
asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml
asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml
asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml
Modified: asterisk/trunk/tests/channels/SIP/alwaysauthreject/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/alwaysauthreject/test-config.yaml?view=diff&rev=4117&r1=4116&r2=4117
==============================================================================
--- asterisk/trunk/tests/channels/SIP/alwaysauthreject/test-config.yaml (original)
+++ asterisk/trunk/tests/channels/SIP/alwaysauthreject/test-config.yaml Thu Sep 5 14:23:32 2013
@@ -4,7 +4,7 @@
"This test verifies that Asterisk responds properly to REGISTER, SUBSCRIBE, and INVITE requests with alwaysauthreject enabled."
properties:
- minversion: '1.8.21.1'
+ minversion: '12.0.0'
dependencies:
- python : 'twisted'
- python : 'starpy'
Modified: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A.xml?view=diff&rev=4117&r1=4116&r2=4117
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A.xml (original)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A.xml Thu Sep 5 14:23:32 2013
@@ -74,7 +74,18 @@
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
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/sip_hold/sipp/phone_B_IP_media_restrict.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml?view=diff&rev=4117&r1=4116&r2=4117
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml (original)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml Thu Sep 5 14:23:32 2013
@@ -61,7 +61,18 @@
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
Content-Type: application/sdp
- Content-Length: 0
+ Content-Length: [len]
+
+ v=0
+ o=- 1325003603 1325003604 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ a=sendonly
+ m=audio 2226 RTP/AVP 0 101
+ a=sendonly
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
]]>
</send>
@@ -200,4 +211,4 @@
</send>
-</scenario>
+</scenario>
Modified: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml?view=diff&rev=4117&r1=4116&r2=4117
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml (original)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml Thu Sep 5 14:23:32 2013
@@ -61,7 +61,18 @@
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
Content-Type: application/sdp
- Content-Length: 0
+ Content-Length: [len]
+
+ v=0
+ o=- 1325003603 1325003604 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ a=sendonly
+ m=audio 2226 RTP/AVP 0 101
+ a=sendonly
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
]]>
</send>
@@ -200,4 +211,4 @@
</send>
-</scenario>
+</scenario>
Modified: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml?view=diff&rev=4117&r1=4116&r2=4117
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml (original)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml Thu Sep 5 14:23:32 2013
@@ -61,7 +61,18 @@
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
Content-Type: application/sdp
- Content-Length: 0
+ Content-Length: [len]
+
+ v=0
+ o=- 1325003603 1325003604 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ a=sendonly
+ m=audio 2226 RTP/AVP 0 101
+ a=sendonly
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
]]>
</send>
@@ -201,4 +212,4 @@
</send>
-</scenario>
+</scenario>
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