[asterisk-commits] kharwell: testsuite/asterisk/trunk r4766 - in /asterisk/trunk/tests/channels/...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Mar 4 09:21:01 CST 2014


Author: kharwell
Date: Tue Mar  4 09:20:58 2014
New Revision: 4766

URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=4766
Log:
res_pjsip_send_to_voicemail: transferring to voicemail for digium phones

Tests the send to voicemail feature module to make sure it appropriately parses
the relevant headers in a REFER and sets the correct channels variables.

Review: https://reviewboard.asterisk.org/r/3246/

Added:
    asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/
    asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/configs/
    asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/configs/ast1/
    asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/configs/ast1/extensions.conf   (with props)
    asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/configs/ast1/pjsip.conf   (with props)
    asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/sipp/
    asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/sipp/invite.xml   (with props)
    asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/sipp/refer.xml   (with props)
    asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/test-config.yaml   (with props)
Modified:
    asterisk/trunk/tests/channels/pjsip/tests.yaml

Added: asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/configs/ast1/extensions.conf?view=auto&rev=4766
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/configs/ast1/extensions.conf Tue Mar  4 09:20:58 2014
@@ -1,0 +1,13 @@
+[general]
+
+[default]
+
+exten => bob,1,Dial(PJSIP/${EXTEN})
+      same => n,Hangup()
+
+exten => carol,1,NoOp()
+      same => n,GotoIf($[${REDIRECTING(reason)} = "send_to_vm" | "${PJSIP_HEADER(read,X-Digium-Call-Feature)}" = "feature_send_to_vm"]?vm:notvm)
+      same => n(vm),UserEvent(Result, Status: passed)
+      same => n,Hangup()
+      same => n(notvm),UserEvent(Result, Status: failed)
+      same => n,Hangup()

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Added: asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/configs/ast1/pjsip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/configs/ast1/pjsip.conf?view=auto&rev=4766
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/configs/ast1/pjsip.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/configs/ast1/pjsip.conf Tue Mar  4 09:20:58 2014
@@ -1,0 +1,48 @@
+[local]
+type=transport
+protocol=udp
+bind=0.0.0.0
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[endpoint_t](!)
+type=endpoint
+transport=local
+context=default
+direct_media=no
+disallow=all
+allow=ulaw
+
+[aor_t](!)
+type=aor
+max_contacts=1
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+;;; alice
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[alice](aor_t)
+contact=sip:alice at 127.0.0.1:5061
+
+[alice](endpoint_t)
+aors=alice
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+;;; bob
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[bob](aor_t)
+contact=sip:bob at 127.0.0.1:5062
+
+[bob](endpoint_t)
+aors=bob
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+;;; carol
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[carol](aor_t)
+contact=sip:carol at 127.0.0.1:5063
+
+[carol](endpoint_t)
+aors=carol

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Added: asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/sipp/invite.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/sipp/invite.xml?view=auto&rev=4766
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/sipp/invite.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/sipp/invite.xml Tue Mar  4 09:20:58 2014
@@ -1,0 +1,69 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="Refer with custom header">
+	<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=[call_number]
+                        To: bob <sip:bob@[remote_ip]:[remote_port]>
+                        Call-ID: [call_id]
+                        CSeq: [cseq] INVITE
+                        Contact: sip:alice@[local_ip]:[local_port]
+                        Max-Forwards: 70
+                        Content-Type: application/sdp
+                        Content-Length: [len]
+
+			v=0
+			o=- 1324901698 1324901698 IN IP4 [local_ip]
+			s=-
+			c=IN IP4 [local_ip]
+			t=0 0
+			m=audio 2226 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="180" optional="true" />
+
+	<recv response="200" />
+
+	<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]>;tag=[call_number]
+			To: bob <sip:bob@[remote_ip]:[remote_port]>[peer_tag_param]
+			CSeq: [cseq] ACK
+			Call-ID: [call_id]
+			Contact: alice <sip:alice@[local_ip]:[local_port]>
+                        Allow: INVITE, ACK, MESSAGE, BYE
+			Max-Forwards: 70
+			Content-Length: 0
+
+		]]>
+	</send>
+
+	<recv request="BYE" crlf="true" />
+
+	<send>
+		<![CDATA[
+			SIP/2.0 200 OK
+			[last_Via:]
+			[last_From:]
+			[last_To:];tag=[call_number]
+			[last_Call-ID:]
+			[last_CSeq:]
+			Contact: <sip:alice@[local_ip]:[local_port];transport=[transport]>
+                        Allow: INVITE, ACK, MESSAGE, BYE
+			Content-Type: application/sdp
+			Content-Length: 0
+
+		]]>
+	</send>
+</scenario>

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Added: asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/sipp/refer.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/sipp/refer.xml?view=auto&rev=4766
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/sipp/refer.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/sipp/refer.xml Tue Mar  4 09:20:58 2014
@@ -1,0 +1,118 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="Refer with custom header">
+  <Global variables="remote_tag" />
+  <recv request="INVITE" crlf="true">
+      <!-- Save the from tag. We'll need it when we send our BYE -->
+      <action>
+          <ereg regexp=".*(;tag=.*)"
+              header="From:"
+              search_in="hdr"
+              check_it="true"
+              assign_to="to_hdr,remote_tag"/>
+	  </action>
+  </recv>
+  <Reference variables="to_hdr" />
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 180 Ringing
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag01[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag01[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=- 1324901698 1324901698 IN IP4 [local_ip]
+      s=-
+      c=IN IP4 [local_ip]
+      t=0 0
+      m=audio 2226 RTP/AVP 0 101
+      a=sendrecv
+      a=rtpmap:0 PCMU/8000
+      a=rtpmap:101 telephone-event/8000
+    ]]>
+  </send>
+
+  <recv request="ACK" rtd="true" crlf="true" />
+
+  <pause milliseconds="1000"/>
+
+  <send retrans="500">
+    <![CDATA[
+      REFER sip:alice@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
+      From: <sip:bob@[local_ip]:[local_port]>;tag=[pid]SIPpTag01[call_number]
+      To: <sip:alice@[remote_ip]:[remote_port]>[$remote_tag]
+      Call-ID: [call_id]
+      CSeq: [cseq] REFER
+      Contact: <sip:bob@[local_ip]:[local_port]>
+      Max-Forwards: 70
+      Event: refer
+      Expires: 600
+      Supported: replaces, 100rel, timer, norefersub
+      Accept: message/sipfrag;version=2.0
+      Allow-Events: presence, message-summary, refer
+      Refer-To: <sip:carol@[remote_ip]:[remote_port]>
+      Referred-By: <sip:bob@[local_ip]:[local_port]>
+      User-Agent: Digium D40
+      X-Digium-Call-Feature: feature_send_to_vm
+      Diversion: <sip:carol@[local_ip]:[local_port]>;reason="send_to_vm"
+      Content-Length: [len]
+    ]]>
+  </send>
+
+  <recv response="202" />
+
+  <recv request="NOTIFY" crlf="true" />
+
+  <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];transport=[transport]>
+      Content-Length: 0
+    ]]>
+  </send>
+
+  <recv request="NOTIFY" crlf="true" />
+
+  <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];transport=[transport]>
+      Content-Length: 0
+    ]]>
+  </send>
+</scenario>

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Added: asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/test-config.yaml?view=auto&rev=4766
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/test-config.yaml Tue Mar  4 09:20:58 2014
@@ -1,0 +1,47 @@
+testinfo:
+    summary: 'Test the send to voicemail headers in a refer.'
+    description: |
+        'When using a Digium phone depending on the configuration it is
+         possible for a REFER to contain a diversion and/or custom header.
+         This tests that the appropriate variables are set on the channel
+         before entering the dialplan when those headers are present in
+         a REFER.'
+
+properties:
+    minversion: '12.2.0'
+    dependencies:
+         - app : 'sipp'
+         - asterisk : 'res_pjsip'
+         - asterisk : 'res_pjsip_refer'
+         - asterisk : 'res_pjsip_send_to_voicemail'
+    tags:
+        - pjsip
+
+test-modules:
+    test-object:
+        config-section: test-object-config
+        typename: 'sipp.SIPpTestCase'
+    modules:
+        -
+            config-section: ami-config
+            typename: 'ami.AMIEventModule'
+
+test-object-config:
+    test-iterations:
+        -
+             scenarios:
+                - { 'key-args': { 'scenario':'refer.xml', '-p':'5062' } }
+                - { 'key-args': { 'scenario':'invite.xml', '-p':'5061' } }
+
+ami-config:
+    -
+        id: '0'
+        type: 'headermatch'
+        count: '1'
+        conditions:
+            match:
+                Event: 'UserEvent'
+        requirements:
+            match:
+                Status: 'passed'
+

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Modified: asterisk/trunk/tests/channels/pjsip/tests.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/tests.yaml?view=diff&rev=4766&r1=4765&r2=4766
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/tests.yaml (original)
+++ asterisk/trunk/tests/channels/pjsip/tests.yaml Tue Mar  4 09:20:58 2014
@@ -17,3 +17,4 @@
     - test: 'presence_pidf'
     - test: 'presence_xpidf'
     - test: 'mwi'
+    - test: 'refer_send_to_vm'




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