[asterisk-commits] kmoore: testsuite/asterisk/trunk r3732 - in /asterisk/trunk/tests/channels/SI...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Apr 22 14:50:38 CDT 2013


Author: kmoore
Date: Mon Apr 22 14:50:34 2013
New Revision: 3732

URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=3732
Log:
Add tests for SIP alwaysauthreject user enumeration

Add tests to ensure that alwaysauthreject behaves properly in several
scenarios. The XML scenarios here were largely derived from scenarios
provided by Walter Doekes when he disclosed the vulnerability.

Added:
    asterisk/trunk/tests/channels/SIP/alwaysauthreject/
    asterisk/trunk/tests/channels/SIP/alwaysauthreject/configs/
    asterisk/trunk/tests/channels/SIP/alwaysauthreject/configs/ast1/
    asterisk/trunk/tests/channels/SIP/alwaysauthreject/configs/ast1/sip.conf   (with props)
    asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/
    asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/invite-1.xml   (with props)
    asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/invite-2.xml   (with props)
    asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/invite-3.xml   (with props)
    asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/register.xml   (with props)
    asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/subscribe.xml   (with props)
    asterisk/trunk/tests/channels/SIP/alwaysauthreject/test-config.yaml   (with props)
Modified:
    asterisk/trunk/tests/channels/SIP/tests.yaml

Added: asterisk/trunk/tests/channels/SIP/alwaysauthreject/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/alwaysauthreject/configs/ast1/sip.conf?view=auto&rev=3732
==============================================================================
--- asterisk/trunk/tests/channels/SIP/alwaysauthreject/configs/ast1/sip.conf (added)
+++ asterisk/trunk/tests/channels/SIP/alwaysauthreject/configs/ast1/sip.conf Mon Apr 22 14:50:34 2013
@@ -1,0 +1,9 @@
+[general]
+allowguest=no
+
+[user](!)
+type=friend
+host=dynamic
+secret=noemptysecret
+
+[alice](user)

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Added: asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/invite-1.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/invite-1.xml?view=auto&rev=3732
==============================================================================
--- asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/invite-1.xml (added)
+++ asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/invite-1.xml Mon Apr 22 14:50:34 2013
@@ -1,0 +1,51 @@
+<?xml version="1.0" encoding="UTF-8" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- Derived from scenarios originally created by Walter Doekes -->
+<scenario name="Asterisk INVITE username disclosure 1">
+  <send retrans="500" start_txn="invite">
+    <![CDATA[
+
+      INVITE sip:321@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-[pid]-1
+      From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
+      To: sip:321@[remote_ip]:[remote_port]
+      Contact: sip:[service]@[local_ip]:[local_port]
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Max-Forwards: 70
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 8 0
+      a=rtpmap:8 PCMA/8000
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true" response_txn="invite"/>
+
+  <recv response="401" response_txn="invite"/>
+
+  <send ack_txn="invite">
+    <![CDATA[
+
+      ACK sip:321@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-[pid]-1
+      [routes]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      CSeq: 1 ACK
+      Max-Forwards: 70
+      Content-Length: 0
+
+    ]]>
+  </send>
+</scenario>

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Added: asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/invite-2.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/invite-2.xml?view=auto&rev=3732
==============================================================================
--- asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/invite-2.xml (added)
+++ asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/invite-2.xml Mon Apr 22 14:50:34 2013
@@ -1,0 +1,103 @@
+<?xml version="1.0" encoding="UTF-8" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- Derived from scenarios originally created by Walter Doekes -->
+<scenario name="Asterisk INVITE username disclosure 2">
+  <send retrans="500" start_txn="invite">
+    <![CDATA[
+
+      INVITE sip:321@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-[pid]-1
+      From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
+      To: sip:321@[remote_ip]:[remote_port]
+      Contact: sip:[service]@[local_ip]:[local_port]
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Max-Forwards: 70
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 8 0
+      a=rtpmap:8 PCMA/8000
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true" response_txn="invite"/>
+
+  <recv response="401" auth="true" response_txn="invite"/>
+
+  <send ack_txn="invite">
+    <![CDATA[
+
+      ACK sip:321@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-[pid]-1
+      [routes]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      CSeq: 1 ACK
+      Max-Forwards: 70
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send retrans="500" start_txn="invite">
+    <![CDATA[
+
+      INVITE sip:321@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-[pid]-2
+      From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
+      To: sip:321@[remote_ip]:[remote_port]
+      Contact: sip:[service]@[local_ip]:[local_port]
+      [authentication]
+      Call-ID: [call_id]
+      CSeq: 2 INVITE
+      Max-Forwards: 70
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 8 0
+      a=rtpmap:8 PCMA/8000
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true" response_txn="invite"/>
+
+  <recv response="403" response_txn="invite">
+    <action>
+      <ereg regexp="SIP.2.0 403 Forbidden[\r\n]" assign_to="dummy" check_it="true"/>
+    </action>
+  </recv>
+  <Reference variables="dummy"/>
+
+  <send ack_txn="invite">
+    <![CDATA[
+
+      ACK sip:321@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-[pid]-2
+      [routes]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      CSeq: 2 ACK
+      Max-Forwards: 70
+      Content-Length: 0
+
+    ]]>
+  </send>
+</scenario>

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Added: asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/invite-3.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/invite-3.xml?view=auto&rev=3732
==============================================================================
--- asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/invite-3.xml (added)
+++ asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/invite-3.xml Mon Apr 22 14:50:34 2013
@@ -1,0 +1,148 @@
+<?xml version="1.0" encoding="UTF-8" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- Derived from scenarios originally created by Walter Doekes -->
+<scenario name="Asterisk INVITE username disclosure 3">
+  <send retrans="500" start_txn="invite">
+    <![CDATA[
+
+      INVITE sip:321@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-[pid]-1
+      From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
+      To: sip:321@[remote_ip]:[remote_port]
+      Contact: sip:[service]@[local_ip]:[local_port]
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Max-Forwards: 70
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 8 0
+      a=rtpmap:8 PCMA/8000
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true" response_txn="invite"/>
+
+  <recv response="401" auth="true" response_txn="invite"/>
+
+  <send ack_txn="invite">
+    <![CDATA[
+
+      ACK sip:321@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-[pid]-1
+      [routes]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      CSeq: 1 ACK
+      Max-Forwards: 70
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send retrans="500" start_txn="invite">
+    <![CDATA[
+
+      INVITE sip:321@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-[pid]-2
+      From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
+      To: sip:321@[remote_ip]:[remote_port]
+      Contact: sip:[service]@[local_ip]:[local_port]
+      [authentication]
+      Call-ID: [call_id]
+      CSeq: 2 INVITE
+      Max-Forwards: 70
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 8 0
+      a=rtpmap:8 PCMA/8000
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true" response_txn="invite"/>
+
+  <recv response="403" response_txn="invite"/>
+
+  <send ack_txn="invite">
+    <![CDATA[
+
+      ACK sip:321@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-[pid]-2
+      [routes]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      CSeq: 1 ACK
+      Max-Forwards: 70
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- Fake a retransmit -->
+
+  <send start_txn="invite">
+    <![CDATA[
+
+      INVITE sip:321@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-[pid]-2
+      From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
+      To: sip:321@[remote_ip]:[remote_port]
+      Contact: sip:[service]@[local_ip]:[local_port]
+      [authentication]
+      Call-ID: [call_id]
+      CSeq: 2 INVITE
+      Max-Forwards: 70
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 8 0
+      a=rtpmap:8 PCMA/8000
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true" response_txn="invite"/>
+
+  <recv response="403" response_txn="invite" timeout="500"/>
+
+  <send ack_txn="invite">
+    <![CDATA[
+
+      ACK sip:321@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-[pid]-2
+      [routes]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      CSeq: 2 ACK
+      Max-Forwards: 70
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+</scenario>

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Added: asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/register.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/register.xml?view=auto&rev=3732
==============================================================================
--- asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/register.xml (added)
+++ asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/register.xml Mon Apr 22 14:50:34 2013
@@ -1,0 +1,62 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- Derived from scenarios originally created by Walter Doekes -->
+<scenario name="Asterisk REGISTER username disclosure">
+  <send retrans="500">
+    <![CDATA[
+
+      REGISTER sip:[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: <sip:[service]@[remote_ip]:[remote_port]>;tag=[pid]SIPpTag00[call_number]
+      To: <sip:[service]@[remote_ip]:[remote_port]>
+      Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
+      Expires: 15
+      Call-ID: [call_id]
+      CSeq: 1 REGISTER
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="401" auth="true"/>
+
+  <send retrans="500">
+    <![CDATA[
+
+      REGISTER sip:[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: <sip:[service]@[remote_ip]:[remote_port]>;tag=[pid]SIPpTag00[call_number]
+      To: <sip:[service]@[remote_ip]:[remote_port]>
+      Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
+      Expires: 15
+      [authentication]
+      Call-ID: [call_id]
+      CSeq: 2 REGISTER
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="403"/>
+
+  <!-- Fake a retransmission -->
+  <send retrans="500">
+    <![CDATA[
+
+      REGISTER sip:[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: <sip:[service]@[remote_ip]:[remote_port]>;tag=[pid]SIPpTag00[call_number]
+      To: <sip:[service]@[remote_ip]:[remote_port]>
+      Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
+      Expires: 15
+      [authentication]
+      Call-ID: [call_id]
+      CSeq: 2 REGISTER
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="401"/>
+</scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->

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Added: asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/subscribe.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/subscribe.xml?view=auto&rev=3732
==============================================================================
--- asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/subscribe.xml (added)
+++ asterisk/trunk/tests/channels/SIP/alwaysauthreject/sipp/subscribe.xml Mon Apr 22 14:50:34 2013
@@ -1,0 +1,69 @@
+<?xml version="1.0" encoding="UTF-8" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- Derived from scenarios originally created by Walter Doekes -->
+<scenario name="Asterisk SUBSCRIBE username disclosure">
+  <send retrans="500" start_txn="invite">
+    <![CDATA[
+
+      SUBSCRIBE sip:321@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-[pid]-1
+      From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
+      To: sip:321@[remote_ip]:[remote_port]
+      Contact: sip:[service]@[local_ip]:[local_port]
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Max-Forwards: 70
+      Event: presence
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+    ]]>
+  </send>
+
+  <recv response="401" auth="true" response_txn="invite"/>
+
+  <send retrans="500" start_txn="invite">
+    <![CDATA[
+
+      SUBSCRIBE sip:321@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-[pid]-2
+      From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
+      To: sip:321@[remote_ip]:[remote_port]
+      Contact: sip:[service]@[local_ip]:[local_port]
+      [authentication]
+      Call-ID: [call_id]
+      CSeq: 2 INVITE
+      Max-Forwards: 70
+      Event: presence
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+    ]]>
+  </send>
+
+  <recv response="403" response_txn="invite"/>
+
+  <!-- Fake a retransmission -->
+  <send retrans="500" start_txn="invite">
+    <![CDATA[
+
+      SUBSCRIBE sip:321@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-[pid]-2
+      From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
+      To: sip:321@[remote_ip]:[remote_port]
+      Contact: sip:[service]@[local_ip]:[local_port]
+      [authentication]
+      Call-ID: [call_id]
+      CSeq: 2 INVITE
+      Max-Forwards: 70
+      Event: presence
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+    ]]>
+  </send>
+
+  <recv response="401" response_txn="invite"/>
+
+</scenario>

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Added: 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=auto&rev=3732
==============================================================================
--- asterisk/trunk/tests/channels/SIP/alwaysauthreject/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/SIP/alwaysauthreject/test-config.yaml Mon Apr 22 14:50:34 2013
@@ -1,0 +1,35 @@
+testinfo:
+    summary:     'Test alwaysauthreject with various transactions'
+    description: |
+        "This test verifies that Asterisk responds properly to REGISTER, SUBSCRIBE, and INVITE requests with alwaysauthreject enabled."
+
+properties:
+    minversion: '1.8.21.1'
+    dependencies:
+        - python : 'twisted'
+        - python : 'starpy'
+        - app : 'sipp'
+    tags:
+        - SIP
+
+test-modules:
+    test-object:
+        config-section: sipp-config
+        typename: 'sipp.SIPpTestCase'
+
+sipp-config:
+    reactor-timeout: 40
+    fail-on-any: True
+    test-iterations:
+        -
+            scenarios:
+                - { 'key-args': {'scenario': 'invite-1.xml',  '-s': 'alice', '-p': '5061'} }
+                - { 'key-args': {'scenario': 'invite-1.xml',  '-s': 'kirk',  '-p': '5062'} }
+                - { 'key-args': {'scenario': 'invite-2.xml',  '-s': 'alice', '-p': '5063'} }
+                - { 'key-args': {'scenario': 'invite-2.xml',  '-s': 'kirk',  '-p': '5064'} }
+                - { 'key-args': {'scenario': 'invite-3.xml',  '-s': 'alice', '-p': '5065'} }
+                - { 'key-args': {'scenario': 'invite-3.xml',  '-s': 'kirk',  '-p': '5066'} }
+                - { 'key-args': {'scenario': 'subscribe.xml', '-s': 'alice', '-p': '5067'} }
+                - { 'key-args': {'scenario': 'subscribe.xml', '-s': 'kirk',  '-p': '5068'} }
+                - { 'key-args': {'scenario': 'register.xml',  '-s': 'alice', '-p': '5069'} }
+                - { 'key-args': {'scenario': 'register.xml',  '-s': 'kirk',  '-p': '5070'} }

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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=3732&r1=3731&r2=3732
==============================================================================
--- asterisk/trunk/tests/channels/SIP/tests.yaml (original)
+++ asterisk/trunk/tests/channels/SIP/tests.yaml Mon Apr 22 14:50:34 2013
@@ -57,3 +57,4 @@
     - test: 'directrtpsetup'
     - dir: 'session_timers'
     - test: 'path'
+    - test: 'alwaysauthreject'




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