[asterisk-commits] testsuite: Move pjsip/transfers/../callee local anonymous t... (testsuite[master])

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu May 11 16:58:53 CDT 2017


George Joseph has submitted this change and it was merged. ( https://gerrit.asterisk.org/5613 )

Change subject: testsuite:  Move pjsip/transfers/../callee_local_anonymous to headers
......................................................................


testsuite:  Move pjsip/transfers/../callee_local_anonymous to headers

We've determined that anonymizing the From header when doing a
callee transfer re-invite is not correct.  There are no other
anonymized From tests though so this one was moved to the "headers"
directory and converted to do just a basic call with
callerid_privacy=prohib set.  The UAS (Bob) verifies that the From
header on the INVITE is anonymized.

A second test was also added to confirm that when
callerid_privacy=prohib is NOT set, the From header isn't
anonymized.

Change-Id: I5b67ff0b7b1909b1db2edcd2f982a6a057c8c12e
---
A tests/channels/pjsip/headers/anonymous_from_basic_call/configs/ast1/extensions.conf
R tests/channels/pjsip/headers/anonymous_from_basic_call/configs/ast1/pjsip.conf
A tests/channels/pjsip/headers/anonymous_from_basic_call/sipp/alice.xml
A tests/channels/pjsip/headers/anonymous_from_basic_call/sipp/bob.xml
A tests/channels/pjsip/headers/anonymous_from_basic_call/test-config.yaml
A tests/channels/pjsip/headers/non-anonymous_from_basic_call/configs/ast1/extensions.conf
A tests/channels/pjsip/headers/non-anonymous_from_basic_call/configs/ast1/pjsip.conf
A tests/channels/pjsip/headers/non-anonymous_from_basic_call/sipp/alice.xml
A tests/channels/pjsip/headers/non-anonymous_from_basic_call/sipp/bob.xml
A tests/channels/pjsip/headers/non-anonymous_from_basic_call/test-config.yaml
A tests/channels/pjsip/headers/tests.yaml
M tests/channels/pjsip/tests.yaml
D tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/extensions.conf
D tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referee.xml
D tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referer_uas.xml
D tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uac-no-hangup.xml
D tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uas.xml
D tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/test-config.yaml
18 files changed, 443 insertions(+), 785 deletions(-)

Approvals:
  Mark Michelson: Looks good to me, approved
  Jenkins2: Approved for Submit
  Joshua Colp: Looks good to me, but someone else must approve



diff --git a/tests/channels/pjsip/headers/anonymous_from_basic_call/configs/ast1/extensions.conf b/tests/channels/pjsip/headers/anonymous_from_basic_call/configs/ast1/extensions.conf
new file mode 100644
index 0000000..bc9969a
--- /dev/null
+++ b/tests/channels/pjsip/headers/anonymous_from_basic_call/configs/ast1/extensions.conf
@@ -0,0 +1,5 @@
+
+[default]
+exten => bob,1,NoOp()
+	same => n,Dial(PJSIP/bob)
+	same => n,Hangup()
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/pjsip.conf b/tests/channels/pjsip/headers/anonymous_from_basic_call/configs/ast1/pjsip.conf
similarity index 64%
rename from tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/pjsip.conf
rename to tests/channels/pjsip/headers/anonymous_from_basic_call/configs/ast1/pjsip.conf
index d7e4874..1b000aa 100644
--- a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/pjsip.conf
+++ b/tests/channels/pjsip/headers/anonymous_from_basic_call/configs/ast1/pjsip.conf
@@ -24,15 +24,5 @@
 
 [bob]
 type=aor
-contact=sip:bob at 127.0.0.1:5066
+contact=sip:bob at 127.0.0.1:5063
 
-[charlie](endpoint)
-aors=charlie
-callerid=Charlie <charlie>
-
-[charlie]
-type=aor
-contact=sip:charlie at 127.0.0.1:5067
-
-[david](endpoint)
-callerid=David <david>
diff --git a/tests/channels/pjsip/headers/anonymous_from_basic_call/sipp/alice.xml b/tests/channels/pjsip/headers/anonymous_from_basic_call/sipp/alice.xml
new file mode 100644
index 0000000..976cbec
--- /dev/null
+++ b/tests/channels/pjsip/headers/anonymous_from_basic_call/sipp/alice.xml
@@ -0,0 +1,79 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic UAC Requestor">
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:bob@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: alice <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: <sip:bob@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:alice@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      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 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="200" rtd="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:bob@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: alice <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: <sip:bob@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: sip:alice@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <pause milliseconds="1000"/>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:bob@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: alice <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: <sip:bob@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 BYE
+      Contact: sip:alice@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200"/>
+
+</scenario>
+
diff --git a/tests/channels/pjsip/headers/anonymous_from_basic_call/sipp/bob.xml b/tests/channels/pjsip/headers/anonymous_from_basic_call/sipp/bob.xml
new file mode 100644
index 0000000..1fffcf1
--- /dev/null
+++ b/tests/channels/pjsip/headers/anonymous_from_basic_call/sipp/bob.xml
@@ -0,0 +1,91 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic UAS responder">
+
+  <recv request="INVITE">
+      <action>
+          <ereg regexp="\"Anonymous\" <sip:anonymous at anonymous.invalid>"
+              header="From"
+              search_in="hdr"
+              check_it="true"
+              assign_to="from"/>
+          <ereg regexp="\"Alice\" <sip:alice at 127.0.0.1>"
+              header="P-Asserted-Identity"
+              search_in="hdr"
+              check_it="true"
+              assign_to="asserted_identity"/>
+          <ereg regexp="\"Alice\" <sip:alice at 127.0.0.1>"
+              header="Remote-Party-ID"
+              search_in="hdr"
+              check_it="true"
+              assign_to="remote_party_id"/>
+      </action>
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 180 Ringing
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send retrans="500">
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      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 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv request="ACK">
+  </recv>
+
+  <recv request="BYE">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <Reference variables="asserted_identity" />
+  <Reference variables="remote_party_id" />
+  <Reference variables="from" />
+
+</scenario>
+
diff --git a/tests/channels/pjsip/headers/anonymous_from_basic_call/test-config.yaml b/tests/channels/pjsip/headers/anonymous_from_basic_call/test-config.yaml
new file mode 100644
index 0000000..12fe669
--- /dev/null
+++ b/tests/channels/pjsip/headers/anonymous_from_basic_call/test-config.yaml
@@ -0,0 +1,30 @@
+testinfo:
+    summary: Test anonymized From headers.
+    description: |
+        Alice calls Bob with callerid_privacy=prohib
+        Bob verifies that the From header is anonymized.
+
+test-modules:
+    test-object:
+        config-section: test-object-config
+        typename: sipp.SIPpTestCase
+
+test-object-config:
+    fail-on-any: True
+    test-iterations:
+        -
+            scenarios:
+                - { 'key-args': {'scenario':'alice.xml', '-p':'5062'} }
+                - { 'key-args': {'scenario':'bob.xml', '-p':'5063'} }
+
+properties:
+    minversion: '13.8.0'
+    dependencies:
+        - python : twisted
+        - python : starpy
+        - asterisk : app_dial
+        - asterisk : chan_pjsip
+        - asterisk : res_pjsip_caller_id
+        - asterisk : res_pjsip_session
+    tags:
+        - pjsip
diff --git a/tests/channels/pjsip/headers/non-anonymous_from_basic_call/configs/ast1/extensions.conf b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/configs/ast1/extensions.conf
new file mode 100644
index 0000000..bc9969a
--- /dev/null
+++ b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/configs/ast1/extensions.conf
@@ -0,0 +1,5 @@
+
+[default]
+exten => bob,1,NoOp()
+	same => n,Dial(PJSIP/bob)
+	same => n,Hangup()
diff --git a/tests/channels/pjsip/headers/non-anonymous_from_basic_call/configs/ast1/pjsip.conf b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..9e29cca
--- /dev/null
+++ b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/configs/ast1/pjsip.conf
@@ -0,0 +1,27 @@
+[local]
+type=transport
+protocol=udp
+bind=127.0.0.1:5060
+
+[endpoint](!)
+type=endpoint
+context=default
+disallow=all
+allow=ulaw
+direct_media=no
+send_pai=yes
+send_rpid=yes
+trust_id_outbound=yes
+trust_id_inbound=yes
+
+[alice](endpoint)
+callerid=Alice <alice>
+
+[bob](endpoint)
+aors=bob
+callerid=Bob <bob>
+
+[bob]
+type=aor
+contact=sip:bob at 127.0.0.1:5063
+
diff --git a/tests/channels/pjsip/headers/non-anonymous_from_basic_call/sipp/alice.xml b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/sipp/alice.xml
new file mode 100644
index 0000000..976cbec
--- /dev/null
+++ b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/sipp/alice.xml
@@ -0,0 +1,79 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic UAC Requestor">
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:bob@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: alice <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: <sip:bob@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:alice@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      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 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="200" rtd="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:bob@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: alice <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: <sip:bob@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: sip:alice@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <pause milliseconds="1000"/>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:bob@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: alice <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: <sip:bob@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 BYE
+      Contact: sip:alice@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200"/>
+
+</scenario>
+
diff --git a/tests/channels/pjsip/headers/non-anonymous_from_basic_call/sipp/bob.xml b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/sipp/bob.xml
new file mode 100644
index 0000000..c05dfab
--- /dev/null
+++ b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/sipp/bob.xml
@@ -0,0 +1,91 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic UAS responder">
+
+  <recv request="INVITE">
+      <action>
+          <ereg regexp="\"Alice\" <sip:alice at 127.0.0.1>"
+              header="From"
+              search_in="hdr"
+              check_it="true"
+              assign_to="from"/>
+          <ereg regexp="\"Alice\" <sip:alice at 127.0.0.1>"
+              header="P-Asserted-Identity"
+              search_in="hdr"
+              check_it="true"
+              assign_to="asserted_identity"/>
+          <ereg regexp="\"Alice\" <sip:alice at 127.0.0.1>"
+              header="Remote-Party-ID"
+              search_in="hdr"
+              check_it="true"
+              assign_to="remote_party_id"/>
+      </action>
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 180 Ringing
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send retrans="500">
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      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 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv request="ACK">
+  </recv>
+
+  <recv request="BYE">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <Reference variables="asserted_identity" />
+  <Reference variables="remote_party_id" />
+  <Reference variables="from" />
+
+</scenario>
+
diff --git a/tests/channels/pjsip/headers/non-anonymous_from_basic_call/test-config.yaml b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/test-config.yaml
new file mode 100644
index 0000000..132dbe5
--- /dev/null
+++ b/tests/channels/pjsip/headers/non-anonymous_from_basic_call/test-config.yaml
@@ -0,0 +1,30 @@
+testinfo:
+    summary: Test non-anonymized From headers.
+    description: |
+        Alice calls Bob without callerid_privacy=prohib
+        Bob verifies that the From header is NOT anonymized.
+
+test-modules:
+    test-object:
+        config-section: test-object-config
+        typename: sipp.SIPpTestCase
+
+test-object-config:
+    fail-on-any: True
+    test-iterations:
+        -
+            scenarios:
+                - { 'key-args': {'scenario':'alice.xml', '-p':'5062'} }
+                - { 'key-args': {'scenario':'bob.xml', '-p':'5063'} }
+
+properties:
+    minversion: '13.8.0'
+    dependencies:
+        - python : twisted
+        - python : starpy
+        - asterisk : app_dial
+        - asterisk : chan_pjsip
+        - asterisk : res_pjsip_caller_id
+        - asterisk : res_pjsip_session
+    tags:
+        - pjsip
diff --git a/tests/channels/pjsip/headers/tests.yaml b/tests/channels/pjsip/headers/tests.yaml
new file mode 100644
index 0000000..7c3b64d
--- /dev/null
+++ b/tests/channels/pjsip/headers/tests.yaml
@@ -0,0 +1,4 @@
+# Enter tests here in the order they should be considered for execution:
+tests:
+    - test: 'anonymous_from_basic_call'
+    - test: 'non-anonymous_from_basic_call'
diff --git a/tests/channels/pjsip/tests.yaml b/tests/channels/pjsip/tests.yaml
index 64b3ec0..6c69ab1 100644
--- a/tests/channels/pjsip/tests.yaml
+++ b/tests/channels/pjsip/tests.yaml
@@ -6,6 +6,7 @@
     - dir: 'configuration'
     - dir: 'dialplan_functions'
     - dir: 'diversion'
+    - dir: 'headers'
     - dir: 'identify'
     - dir: 'message'
     - dir: 'nat'
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/extensions.conf b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/extensions.conf
deleted file mode 100644
index a299118..0000000
--- a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/configs/ast1/extensions.conf
+++ /dev/null
@@ -1,9 +0,0 @@
-
-[default]
-exten => call_c,1,NoOp()
-	same => n,Dial(PJSIP/charlie)
-	same => n,Hangup()
-
-exten => alice,1,NoOp()
-	same => n,Dial(PJSIP/bob)
-	same => n,Hangup()
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referee.xml b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referee.xml
deleted file mode 100644
index f4166e1..0000000
--- a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referee.xml
+++ /dev/null
@@ -1,137 +0,0 @@
-<?xml version="1.0" encoding="ISO-8859-1" ?>
-<!DOCTYPE scenario SYSTEM "sipp.dtd">
-
-<!-- This program is free software; you can redistribute it and/or      -->
-<!-- modify it under the terms of the GNU General Public License as     -->
-<!-- published by the Free Software Foundation; either version 2 of the -->
-<!-- License, or (at your option) any later version.                    -->
-<!--                                                                    -->
-<!-- This program is distributed in the hope that it will be useful,    -->
-<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
-<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
-<!-- GNU General Public License for more details.                       -->
-<!--                                                                    -->
-<!-- You should have received a copy of the GNU General Public License  -->
-<!-- along with this program; if not, write to the                      -->
-<!-- Free Software Foundation, Inc.,                                    -->
-<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
-<!--                                                                    -->
-
-<scenario name="Referee Leg">
-
-  <recvCmd>
-    <action>
-      <ereg regexp="REMOTE(.*)"
-        search_in="hdr"
-        header="Call-ID:"
-        check_it="true"
-        assign_to="1,original_callid" />
-    </action>
-  </recvCmd>
-
-  <send retrans="500">
-    <![CDATA[
-
-      INVITE sip:call_c@[remote_ip]:[remote_port] SIP/2.0
-      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
-      From: <sip:bob@[local_ip]:[local_port]>;tag=[call_number]
-      To: <sip:transfer@[remote_ip]:[remote_port]>
-      Call-ID: [call_id]
-      CSeq: [cseq] INVITE
-      Contact: <sip:bob@[local_ip]:[local_port]>
-      Max-Forwards: 70
-      Content-Type: application/sdp
-      Content-Length: [len]
-
-      v=0
-      o=- 1324901698 1324901698 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 0 101
-      a=sendrecv
-      a=rtpmap:0 PCMU/8000
-      a=rtpmap:101 telephone-event/8000
-
-    ]]>
-  </send>
-
-  <recv response="100" optional="true" />
-  <recv response="101" optional="true" />
-  <recv response="180" optional="true" />
-  <recv response="200" rtd="true" crlf="true">
-    <action>
-      <ereg regexp="tag=([[:alnum:].\-]*)"
-        search_in="hdr"
-        header="To:"
-        check_it="true"
-        assign_to="2,to_tag" />
-      <ereg regexp="tag=([[:alnum:].\-]*)"
-        search_in="hdr"
-        header="From:"
-        check_it="true"
-        assign_to="3,from_tag" />
-    </action>
-  </recv>
-  <Reference variables="1,2,3" />
-
-  <send>
-    <![CDATA[
-
-      ACK sip:call_c@[remote_ip]:[remote_port] SIP/2.0
-      Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
-      [last_From:]
-      [last_To]
-      Call-ID: [call_id]
-      CSeq: [cseq] ACK
-      Contact: sip:bob@[local_ip]:[local_port]
-      Max-Forwards: 70
-      Content-Length: 0
-
-    ]]>
-  </send>
-
-  <pause milliseconds="1000" />
-  <sendCmd>
-    <![CDATA[
-      Call-ID: [$original_callid]
-      Remote-To-Tag: [$to_tag]
-      Remote-From-Tag: [$from_tag]
-      Remote-URI: sip:call_c@[remote_ip]:[remote_port]
-    ]]>
-  </sendCmd>
-
-  <recv request="BYE">
-    <action>
-        <ereg regexp="<sip:transfer at 127.0.0.1>"
-         header="From"
-         search_in="hdr"
-         check_it="true"
-         assign_to="from"/>
-	</action>
-  </recv>
-  <Reference variables="from" />
-
-  <send>
-    <![CDATA[
-
-      SIP/2.0 200 OK
-      [last_Via:]
-      [last_From:]
-      [last_To]
-      [last_Call-ID:]
-      [last_CSeq:]
-      Contact: <sip:bob@[local_ip]:[local_port]>
-      Content-Length:0
-
-    ]]>
-  </send>
-
-  <!-- definition of the response time repartition table (unit is ms)   -->
-  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
-
-  <!-- definition of the call length repartition table (unit is ms)     -->
-  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
-
-</scenario>
-
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referer_uas.xml b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referer_uas.xml
deleted file mode 100644
index 950cc54..0000000
--- a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/referer_uas.xml
+++ /dev/null
@@ -1,252 +0,0 @@
-<?xml version="1.0" encoding="ISO-8859-1" ?>
-<!DOCTYPE scenario SYSTEM "sipp.dtd">
-
-<!-- This program is free software; you can redistribute it and/or      -->
-<!-- modify it under the terms of the GNU General Public License as     -->
-<!-- published by the Free Software Foundation; either version 2 of the -->
-<!-- License, or (at your option) any later version.                    -->
-<!--                                                                    -->
-<!-- This program is distributed in the hope that it will be useful,    -->
-<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
-<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
-<!-- GNU General Public License for more details.                       -->
-<!--                                                                    -->
-<!-- You should have received a copy of the GNU General Public License  -->
-<!-- along with this program; if not, write to the                      -->
-<!-- Free Software Foundation, Inc.,                                    -->
-<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
-<!--                                                                    -->
-
-<scenario name="Referer Leg">
-  <recv request="INVITE" crlf="true">
-    <action>
-		<ereg regexp="\"Anonymous\" <sip:anonymous at anonymous.invalid>"
-              header="From"
-              search_in="hdr"
-              check_it="true"
-              assign_to="from"/>
-    </action>
-  </recv>
-
-  <send retrans="500">
-    <![CDATA[
-
-      SIP/2.0 200 OK
-      [last_Via:]
-      [last_From:]
-      [last_To:];tag=[call_number]
-      [last_Call-ID:]
-      [last_CSeq:]
-      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
-      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 0
-      a=rtpmap:0 PCMU/8000
-
-    ]]>
-  </send>
-
-  <recv request="ACK"
-        rtd="true"
-        crlf="true">
-    <action>
-      <ereg regexp="\"Anonymous\" <sip:anonymous at anonymous.invalid>"
-         header="From"
-         search_in="hdr"
-         check_it="true"
-         assign_to="from"/>
-      <ereg regexp=" (.+)"
-        search_in="hdr"
-        header="From:"
-        check_it="true"
-        assign_to="1,outbound_to_header" />
-      <ereg regexp=" (.+)"
-        search_in="hdr"
-        header="To:"
-        check_it="true"
-        assign_to="1,outbound_from_header" />
-    </action>
-  </recv>
-
-  <!-- Put this leg on hold -->
-  <send retrans="500">
-    <![CDATA[
-
-      INVITE sip:[remote_ip]:[remote_port] SIP/2.0
-      Via: SIP/2.0/UDP [local_ip]:[local_port];rport;received=127.0.0.1;branch=[branch]
-      From: [$outbound_from_header]
-      To: [$outbound_to_header]
-      Call-ID: [call_id]
-      CSeq: [cseq] INVITE
-      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
-      Content-Type: application/sdp
-      Max-Forwards: 70
-      Content-Length: [len]
-
-      v=0
-      o=- 1324901698 1324901698 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 0 101
-      a=sendonly
-      a=rtpmap:0 PCMU/8000
-      a=rtpmap:101 telephone-event/8000
-
-    ]]>
-  </send>
-
-  <recv response="100" optional="true" />
-  <recv response="101" optional="true" />
-  <recv response="180" optional="true" />
-  <recv response="200" rtd="true" crlf="true" />
-
-  <send>
-    <![CDATA[
-
-      ACK sip:[local_ip]:[local_port] SIP/2.0
-      [last_Via]
-      [last_From]
-      [last_To]
-      Call-ID: [call_id]
-      CSeq: [cseq] ACK
-      Contact: sip:bob@[local_ip]:[local_port]
-      Max-Forwards: 70
-      Content-Length: 0
-
-    ]]>
-  </send>
-
-  <sendCmd>
-    <![CDATA[
-      Call-ID: REMOTE[call_id]
-      Start the Echo Leg
-    ]]>
-  </sendCmd>
-
-  <recvCmd>
-    <action>
-      <ereg regexp=" (.+)"
-        search_in="hdr"
-        header="Remote-URI:"
-        check_it="true"
-        assign_to="1,remote_contact" />
-      <ereg regexp=" (.+)"
-        search_in="hdr"
-        header="Remote-To-Tag:"
-        check_it="true"
-        assign_to="2,remote_to_tag" />
-      <ereg regexp=" (.+)"
-        search_in="hdr"
-        header="Remote-From-Tag:"
-        check_it="true"
-        assign_to="3,remote_from_tag" />
-     </action>
-  </recvCmd>
-  <Reference variables="1,2,3" />
-
-  <send>
-    <![CDATA[
-
-      REFER sip:call_c@[remote_ip]:[remote_port] SIP/2.0
-      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
-      [last_From:]
-      [last_To]
-      [last_Call-ID:]
-      CSeq: [cseq] REFER
-      Contact: <sip:bob@[local_ip]:[local_port]>
-      Max-Forwards: 70
-      Refer-to: <[$remote_contact]?Replaces=REMOTE[call_id]%3Bto-tag%3D[$remote_to_tag]%3Bfrom-tag%3D[$remote_from_tag]>
-      Referred-By: sip:bob@[local_ip]
-      Content-Length: 0
-
-    ]]>
-  </send>
-  <recv response="202" rtd="true" crlf="true" />
-
-  <!-- In a nominal attended transfer Asterisk should always
-       be sending two notifies (SIP frags of 100 and 200) -->
-  <recv request="NOTIFY" >
-    <action>
-        <ereg regexp="\"Anonymous\" <sip:anonymous at anonymous.invalid>"
-         header="From"
-         search_in="hdr"
-         check_it="true"
-         assign_to="from"/>
-    </action>
-  </recv>
-  <send>
-    <![CDATA[
-
-      SIP/2.0 200 OK
-      [last_Via:]
-      [last_From:]
-      [last_To]
-      [last_Call-ID:]
-      [last_CSeq:]
-      Contact: <sip:bob@[local_ip]:[local_port]>
-      Content-Length:0
-
-    ]]>
-  </send>
-
-  <recv request="NOTIFY">
-    <action>
-        <ereg regexp="\"Anonymous\" <sip:anonymous at anonymous.invalid>"
-         header="From"
-         search_in="hdr"
-         check_it="true"
-         assign_to="from"/>
-    </action>
-  </recv>
-  <send>
-    <![CDATA[
-
-      SIP/2.0 200 OK
-      [last_Via:]
-      [last_From:]
-      [last_To]
-      [last_Call-ID:]
-      [last_CSeq:]
-      Contact: <sip:bob@[local_ip]:[local_port]>
-      Content-Length:0
-
-    ]]>
-  </send>
-
-  <send retrans="500">
-    <![CDATA[
-
-      BYE sip:call_c@[remote_ip]:[remote_port] SIP/2.0
-      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
-      From: <sip:bob@[local_ip]:[local_port]>;tag=[call_number]
-      To: <sip:transfer@[remote_ip]:[remote_port]>[peer_tag_param]
-      Call-ID: [call_id]
-      CSeq: [cseq] BYE
-      Contact: sip:bob@[local_ip]:[local_port]
-      Max-Forwards: 70
-      Subject: Performance Test
-      Content-Length: 0
-
-    ]]>
-  </send>
-
-  <recv response="200"/>
-
-  <!-- definition of the response time repartition table (unit is ms)   -->
-  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
-
-  <!-- definition of the call length repartition table (unit is ms)     -->
-  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
-
-  <Reference variables="from" />
-
-</scenario>
-
-
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uac-no-hangup.xml b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uac-no-hangup.xml
deleted file mode 100644
index 321e53f..0000000
--- a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uac-no-hangup.xml
+++ /dev/null
@@ -1,141 +0,0 @@
-<?xml version="1.0" encoding="ISO-8859-1" ?>
-<!DOCTYPE scenario SYSTEM "sipp.dtd">
-
-<scenario name="Basic Sipstone UAC">
-  <send retrans="500">
-    <![CDATA[
-
-      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
-      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
-      From: sipp <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
-      To: sut <sip:[service]@[remote_ip]:[remote_port]>
-      Call-ID: [call_id]
-      CSeq: 1 INVITE
-      Contact: sip:alice@[local_ip]:[local_port]
-      Max-Forwards: 70
-      Subject: Performance Test
-      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 0
-      a=rtpmap:0 PCMU/8000
-
-    ]]>
-  </send>
-
-  <recv response="100"
-        optional="true">
-  </recv>
-
-  <recv response="181"
-        optional="true">
-  </recv>
-
-  <recv response="180" optional="true">
-  </recv>
-
-  <recv response="183" optional="true">
-  </recv>
-
-  <recv response="200" rtd="true">
-  </recv>
-
-  <send>
-    <![CDATA[
-
-      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
-      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
-      From: sipp <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
-      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
-      Call-ID: [call_id]
-      CSeq: 1 ACK
-      Contact: sip:alice@[local_ip]:[local_port]
-      Max-Forwards: 70
-      Subject: Performance Test
-      Content-Length: 0
-
-    ]]>
-  </send>
-
-  <recv request="INVITE">
-      <action>
-          <ereg regexp="\"Anonymous\" <sip:anonymous at anonymous.invalid>"
-              header="From"
-              search_in="hdr"
-              check_it="true"
-              assign_to="from"/>
-          <ereg regexp="\"Charlie\" <sip:charlie at 127.0.0.1>"
-              header="P-Asserted-Identity"
-              search_in="hdr"
-              check_it="true"
-              assign_to="asserted_identity"/>
-          <ereg regexp="\"Charlie\" <sip:charlie at 127.0.0.1>"
-              header="Remote-Party-ID"
-              search_in="hdr"
-              check_it="true"
-              assign_to="remote_party_id"/>
-      </action>
-  </recv>
-
-  <send>
-    <![CDATA[
-
-      SIP/2.0 200 OK
-      [last_Via:]
-      [last_From:]
-      [last_To:]
-      [last_Call-ID:]
-      [last_CSeq:]
-      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
-      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 0
-      a=rtpmap:0 PCMU/8000
-
-    ]]>
-  </send>
-
-  <recv request="ACK">
-  </recv>
-
-  <recv request="BYE">
-  </recv>
-
-  <send>
-    <![CDATA[
-
-      SIP/2.0 200 OK
-      [last_Via:]
-      [last_From:]
-      [last_To:]
-      [last_Call-ID:]
-      [last_CSeq:]
-      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
-      Content-Length: 0
-
-    ]]>
-  </send>
-
-  <timewait milliseconds="4000"/>
-
-  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
-
-  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
-
-  <Reference variables="asserted_identity" />
-  <Reference variables="remote_party_id" />
-  <Reference variables="from" />
-
-</scenario>
-
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uas.xml b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uas.xml
deleted file mode 100644
index 69c014d..0000000
--- a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/sipp/uas.xml
+++ /dev/null
@@ -1,166 +0,0 @@
-<?xml version="1.0" encoding="ISO-8859-1" ?>
-<!DOCTYPE scenario SYSTEM "sipp.dtd">
-
-<!-- This program is free software; you can redistribute it and/or      -->
-<!-- modify it under the terms of the GNU General Public License as     -->
-<!-- published by the Free Software Foundation; either version 2 of the -->
-<!-- License, or (at your option) any later version.                    -->
-<!--                                                                    -->
-<!-- This program is distributed in the hope that it will be useful,    -->
-<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
-<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
-<!-- GNU General Public License for more details.                       -->
-<!--                                                                    -->
-<!-- You should have received a copy of the GNU General Public License  -->
-<!-- along with this program; if not, write to the                      -->
-<!-- Free Software Foundation, Inc.,                                    -->
-<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
-<!--                                                                    -->
-<!--                 Sipp default 'uas' scenario.                       -->
-<!--                                                                    -->
-
-<scenario name="Basic UAS responder">
-  <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
-  <!-- are saved and used for following messages sent. Useful to test   -->
-  <!-- against stateful SIP proxies/B2BUAs.                             -->
-  <recv request="INVITE" crlf="true">
-  </recv>
-
-  <!-- The '[last_*]' keyword is replaced automatically by the          -->
-  <!-- specified header if it was present in the last message received  -->
-  <!-- (except if it was a retransmission). If the header was not       -->
-  <!-- present or if no message has been received, the '[last_*]'       -->
-  <!-- keyword is discarded, and all bytes until the end of the line    -->
-  <!-- are also discarded.                                              -->
-  <!--                                                                  -->
-  <!-- If the specified header was present several times in the         -->
-  <!-- message, all occurences are concatenated (CRLF seperated)        -->
-  <!-- to be used in place of the '[last_*]' keyword.                   -->
-
-  <send>
-    <![CDATA[
-
-      SIP/2.0 180 Ringing
-      [last_Via:]
-      [last_From:]
-      [last_To:];tag=[call_number]
-      [last_Call-ID:]
-      [last_CSeq:]
-      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
-      Content-Length: 0
-
-    ]]>
-  </send>
-
-  <send retrans="500">
-    <![CDATA[
-
-      SIP/2.0 200 OK
-      [last_Via:]
-      [last_From:]
-      [last_To:];tag=[call_number]
-      [last_Call-ID:]
-      [last_CSeq:]
-      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
-      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 0
-      a=rtpmap:0 PCMU/8000
-
-    ]]>
-  </send>
-
-  <recv request="ACK"
-        optional="true"
-        rtd="true"
-        crlf="true">
-  </recv>
-
-  <recv request="INVITE">
-      <action>
-          <ereg regexp="\"Anonymous\" <sip:anonymous at anonymous.invalid>"
-              header="From"
-              search_in="hdr"
-              check_it="true"
-              assign_to="from"/>
-          <ereg regexp="\"Alice\" <sip:alice at 127.0.0.1>"
-              header="P-Asserted-Identity"
-              search_in="hdr"
-              check_it="true"
-              assign_to="asserted_identity"/>
-          <ereg regexp="\"Alice\" <sip:alice at 127.0.0.1>"
-              header="Remote-Party-ID"
-              search_in="hdr"
-              check_it="true"
-              assign_to="remote_party_id"/>
-      </action>
-  </recv>
-
-  <send>
-    <![CDATA[
-
-      SIP/2.0 200 OK
-      [last_Via:]
-      [last_From:]
-      [last_To:]
-      [last_Call-ID:]
-      [last_CSeq:]
-      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
-      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 0
-      a=rtpmap:0 PCMU/8000
-
-    ]]>
-  </send>
-
-  <recv request="ACK">
-  </recv>
-
-  <recv request="BYE">
-  </recv>
-
-  <send>
-    <![CDATA[
-
-      SIP/2.0 200 OK
-      [last_Via:]
-      [last_From:]
-      [last_To:]
-      [last_Call-ID:]
-      [last_CSeq:]
-      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
-      Content-Length: 0
-
-    ]]>
-  </send>
-
-  <!-- Keep the call open for a while in case the 200 is lost to be     -->
-  <!-- able to retransmit it if we receive the BYE again.               -->
-  <pause milliseconds="4000"/>
-
-
-  <!-- definition of the response time repartition table (unit is ms)   -->
-  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
-
-  <!-- definition of the call length repartition table (unit is ms)     -->
-  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
-
-  <Reference variables="asserted_identity" />
-  <Reference variables="remote_party_id" />
-  <Reference variables="from" />
-
-</scenario>
-
diff --git a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/test-config.yaml b/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/test-config.yaml
deleted file mode 100644
index 160148b..0000000
--- a/tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local_anonymous/test-config.yaml
+++ /dev/null
@@ -1,69 +0,0 @@
-testinfo:
-    summary: Test anonymized From headers while performing a callee-initiated attended transfer.
-    description: |
-        "Start four SIPp scenarios that do the following:
-        SIPp #1 (uac-no-hangup.xml) calls through Asterisk to SIPp #2 (referer_uas.xml)
-        SIPp #2 kicks off SIPp #3 (referee.xml) which calls SIPp #4 (uas.xml).
-        SIPp #3 passes call information back to SIPp #2.
-        SIPp #2 initiates an attended transfer via REFER with Replaces information from SIPp #3.
-        SIPp #1 and SIPp #4 are bridged.
-        SIPp #1 and SIPp #4 receive connected line updates and the values are checked.
-        SIPp #2 and SIPp #3 are hung up.
-        SIPp #1 and SIPp #4 are hung up."
-
-test-modules:
-    test-object:
-        config-section: test-object-config
-        typename: sipp.SIPpTestCase
-    modules:
-        -
-            config-section: ami-config
-            typename: 'pluggable_modules.EventActionModule'
-
-test-object-config:
-    fail-on-any: True
-    test-iterations:
-        -
-            scenarios:
-                - { 'coordinated-sender': {'key-args': {'scenario':'referer_uas.xml', '-p':'5066', '-sleep': '2'} },
-                    'coordinated-receiver': { 'key-args': {'scenario':'referee.xml', '-p':'5065'} } }
-                - { 'key-args': {'scenario':'uas.xml', '-p':'5067', '-sleep': '2'} }
-                - { 'key-args': {'scenario':'uac-no-hangup.xml', '-p':'5068', '-s':'alice', '-sleep': '2'} }
-
-ami-config:
-    -
-        ami-events:
-            type: 'headermatch'
-            conditions:
-                match:
-                    Event: 'AttendedTransfer'
-                    Result: 'Success'
-            count: 1
-    # Ensure COLP updates occur for alice and charlie before hanging up.
-    -
-        ami-events:
-            conditions:
-                match:
-                    Event: 'NewConnectedLine'
-                    Channel: 'PJSIP/charlie-.*|PJSIP/alice-.*'
-                    ChannelStateDesc: 'Up'
-                    ConnectedLineNum: 'alice|charlie'
-                    ConnectedLineName: 'Alice|Charlie'
-            count: '>2'
-            trigger-on-count: True
-        ami-actions:
-            action:
-                action: 'Hangup'
-                channel: '/^PJSIP/charlie-.*$/'
-
-properties:
-    minversion: '13.8.0'
-    dependencies:
-        - python : twisted
-        - python : starpy
-        - asterisk : app_dial
-        - asterisk : chan_pjsip
-        - asterisk : res_pjsip_caller_id
-        - asterisk : res_pjsip_session
-    tags:
-        - pjsip

-- 
To view, visit https://gerrit.asterisk.org/5613
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-MessageType: merged
Gerrit-Change-Id: I5b67ff0b7b1909b1db2edcd2f982a6a057c8c12e
Gerrit-PatchSet: 1
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>



More information about the asterisk-commits mailing list