[asterisk-commits] seanbright: branch group/pimp_my_sip r3782 - in /asterisk/team/group/pimp_my_...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri May 17 11:21:58 CDT 2013


Author: seanbright
Date: Fri May 17 11:21:52 2013
New Revision: 3782

URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=3782
Log:
Pimp my SIP Testsuite: One Touch Recording tests

Tests for the One Touch Recording functionality from review #2541.

(issue ASTERISK-21501)
Review: https://reviewboard.asterisk.org/r/2542/

Added:
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/extensions.conf   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/features.conf   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/modules.conf   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/res_sip.conf   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/sipp/
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/sipp/config_default.xml   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/sipp/config_disabled.xml   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/sipp/config_enabled.xml   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/test-config.yaml   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/extensions.conf   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/features.conf   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/modules.conf   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/res_sip.conf   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/sipp/
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/sipp/default.xml   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/test-config.yaml   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_not_configured/
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_not_configured/configs/
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_not_configured/configs/ast1/
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_not_configured/configs/ast1/extensions.conf   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_not_configured/configs/ast1/modules.conf   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_not_configured/configs/ast1/res_sip.conf   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_not_configured/sipp/
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_not_configured/sipp/default.xml   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_not_configured/test-config.yaml   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/tests.yaml   (with props)
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/tests.yaml   (with props)
Modified:
    asterisk/team/group/pimp_my_sip/tests/channels/gulp/tests.yaml

Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/extensions.conf?view=auto&rev=3782
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/extensions.conf (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/extensions.conf Fri May 17 11:21:52 2013
@@ -1,0 +1,4 @@
+[default]
+exten => wait,1,Answer()
+same  =>      n,Wait(30)
+same  =>      n,Hangup()

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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/features.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/features.conf?view=auto&rev=3782
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/features.conf (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/features.conf Fri May 17 11:21:52 2013
@@ -1,0 +1,3 @@
+[featuremap]
+automixmon => *3
+

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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/modules.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/modules.conf?view=auto&rev=3782
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/modules.conf (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/modules.conf Fri May 17 11:21:52 2013
@@ -1,0 +1,19 @@
+[modules]
+autoload = no
+preload => res_sorcery_config.so
+preload => res_sorcery_memory.so
+preload => res_sorcery_astdb.so
+load => res_sip.so
+load => res_sip_logger.so
+load => res_sip_session.so
+load => res_sip_sdp_rtp.so
+load => res_sip_endpoint_identifier_user.so
+load => res_sip_one_touch_record_info.so
+load => res_rtp_asterisk.so
+load => app_playback.so
+load => app_echo.so
+load => format_gsm.so
+load => codec_gsm.so
+load => codec_ulaw.so
+load => pbx_config.so
+load => chan_gulp.so

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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/res_sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/res_sip.conf?view=auto&rev=3782
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/res_sip.conf (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/configs/ast1/res_sip.conf Fri May 17 11:21:52 2013
@@ -1,0 +1,19 @@
+[local-transport-template](!)
+type=transport
+bind=127.0.0.1
+
+[local-transport-udp](local-transport-template)
+protocol=udp
+
+[endpoint-template](!)
+type=endpoint
+context=default
+allow=!all,ulaw,alaw
+
+[alice](endpoint-template)
+
+[bob](endpoint-template)
+one_touch_recording=yes
+
+[carol](endpoint-template)
+one_touch_recording=no

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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/sipp/config_default.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/sipp/config_default.xml?view=auto&rev=3782
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/sipp/config_default.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/sipp/config_default.xml Fri May 17 11:21:52 2013
@@ -1,0 +1,113 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="SIP INFO Record Header Test">
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:wait@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Subject: SIP INFO Record Header Test
+      User-Agent: Test
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio 6000 RTP/AVP 0 3
+      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:wait@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Subject: SIP INFO Record Header Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <pause milliseconds="2500"/>
+
+  <!--
+    Example INFO packet with Record header from:
+    http://wiki.snom.com/Category:HowTo:Call_Recording
+  -->
+
+  <send retrans="500">
+    <![CDATA[
+
+      INFO sip:wait@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 INFO
+      Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Record: on
+      Content-Length: 0
+
+    ]]>
+
+  </send>
+
+  <recv response="403">
+  </recv>
+
+  <pause milliseconds="2500"/>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:wait@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 3 BYE
+      Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Subject: SIP INFO Record Header Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>

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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/sipp/config_disabled.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/sipp/config_disabled.xml?view=auto&rev=3782
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/sipp/config_disabled.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/sipp/config_disabled.xml Fri May 17 11:21:52 2013
@@ -1,0 +1,113 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="SIP INFO Record Header Test">
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:wait@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:carol@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Subject: SIP INFO Record Header Test
+      User-Agent: Test
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio 6000 RTP/AVP 0 3
+      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:wait@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:carol@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Subject: SIP INFO Record Header Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <pause milliseconds="2500"/>
+
+  <!--
+    Example INFO packet with Record header from:
+    http://wiki.snom.com/Category:HowTo:Call_Recording
+  -->
+
+  <send retrans="500">
+    <![CDATA[
+
+      INFO sip:wait@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:carol@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 INFO
+      Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Record: on
+      Content-Length: 0
+
+    ]]>
+
+  </send>
+
+  <recv response="403">
+  </recv>
+
+  <pause milliseconds="2500"/>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:wait@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:carol@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 3 BYE
+      Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Subject: SIP INFO Record Header Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>

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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/sipp/config_enabled.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/sipp/config_enabled.xml?view=auto&rev=3782
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/sipp/config_enabled.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/sipp/config_enabled.xml Fri May 17 11:21:52 2013
@@ -1,0 +1,113 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="SIP INFO Record Header Test">
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:wait@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:bob@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Subject: SIP INFO Record Header Test
+      User-Agent: Test
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio 6000 RTP/AVP 0 3
+      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:wait@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:bob@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Subject: SIP INFO Record Header Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <pause milliseconds="2500"/>
+
+  <!--
+    Example INFO packet with Record header from:
+    http://wiki.snom.com/Category:HowTo:Call_Recording
+  -->
+
+  <send retrans="500">
+    <![CDATA[
+
+      INFO sip:wait@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:bob@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 INFO
+      Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Record: on
+      Content-Length: 0
+
+    ]]>
+
+  </send>
+
+  <recv response="200">
+  </recv>
+
+  <pause milliseconds="2500"/>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:wait@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:bob@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 3 BYE
+      Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Subject: SIP INFO Record Header Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>

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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/test-config.yaml?view=auto&rev=3782
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/test-config.yaml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/endpoint_config/test-config.yaml Fri May 17 11:21:52 2013
@@ -1,0 +1,28 @@
+testinfo:
+    summary:     'Tests One Touch Recording endpoint configuration.'
+    description: |
+        'Run several SIPp scenarios that test the one_touch_recording endpoint
+        configuration options.  The scenarios test endpoints that use the
+        default configuration, explicitly enabled configuration, and explicitly
+        disabled configuration.'
+
+test-modules:
+    test-object:
+        config-section: test-object-config
+        typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+    test-iterations:
+        -
+            scenarios:
+                - { 'key-args': {'scenario': 'config_default.xml', '-i': '127.0.0.1', '-p': '5061'} }
+                - { 'key-args': {'scenario': 'config_enabled.xml', '-i': '127.0.0.1', '-p': '5062'} }
+                - { 'key-args': {'scenario': 'config_disabled.xml', '-i': '127.0.0.1', '-p': '5063'} }
+
+properties:
+    minversion: '12.0.0'
+    dependencies:
+        - sipp :
+            version : 'v3.0'
+    tags:
+        - gulp

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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/extensions.conf?view=auto&rev=3782
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/extensions.conf (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/extensions.conf Fri May 17 11:21:52 2013
@@ -1,0 +1,4 @@
+[default]
+exten => wait,1,Answer()
+same  =>      n,Wait(30)
+same  =>      n,Hangup()

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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/features.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/features.conf?view=auto&rev=3782
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/features.conf (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/features.conf Fri May 17 11:21:52 2013
@@ -1,0 +1,2 @@
+[featuremap]
+automixmon => *8675390

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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/modules.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/modules.conf?view=auto&rev=3782
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/modules.conf (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/modules.conf Fri May 17 11:21:52 2013
@@ -1,0 +1,19 @@
+[modules]
+autoload = no
+preload => res_sorcery_config.so
+preload => res_sorcery_memory.so
+preload => res_sorcery_astdb.so
+load => res_sip.so
+load => res_sip_logger.so
+load => res_sip_session.so
+load => res_sip_sdp_rtp.so
+load => res_sip_endpoint_identifier_user.so
+load => res_sip_one_touch_record_info.so
+load => res_rtp_asterisk.so
+load => app_playback.so
+load => app_echo.so
+load => format_gsm.so
+load => codec_gsm.so
+load => codec_ulaw.so
+load => pbx_config.so
+load => chan_gulp.so

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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/res_sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/res_sip.conf?view=auto&rev=3782
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/res_sip.conf (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/configs/ast1/res_sip.conf Fri May 17 11:21:52 2013
@@ -1,0 +1,14 @@
+[local-transport-template](!)
+type=transport
+bind=127.0.0.1
+
+[local-transport-udp](local-transport-template)
+protocol=udp
+
+[endpoint-template](!)
+type=endpoint
+context=default
+allow=!all,ulaw,alaw
+
+[alice](endpoint-template)
+one_touch_recording=yes

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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/sipp/default.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/sipp/default.xml?view=auto&rev=3782
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/sipp/default.xml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/sipp/default.xml Fri May 17 11:21:52 2013
@@ -1,0 +1,113 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="SIP INFO Record Header Test">
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:wait@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Subject: SIP INFO Record Header Test
+      User-Agent: Test
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio 6000 RTP/AVP 0 3
+      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:wait@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Subject: SIP INFO Record Header Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <pause milliseconds="2500"/>
+
+  <!--
+    Example INFO packet with Record header from:
+    http://wiki.snom.com/Category:HowTo:Call_Recording
+  -->
+
+  <send retrans="500">
+    <![CDATA[
+
+      INFO sip:wait@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 INFO
+      Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Record: on
+      Content-Length: 0
+
+    ]]>
+
+  </send>
+
+  <recv response="200">
+  </recv>
+
+  <pause milliseconds="2500"/>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:wait@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 3 BYE
+      Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+      Max-Forwards: 70
+      Subject: SIP INFO Record Header Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>

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Added: asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/test-config.yaml?view=auto&rev=3782
==============================================================================
--- asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/test-config.yaml (added)
+++ asterisk/team/group/pimp_my_sip/tests/channels/gulp/one_touch_recording/features_config/features_configured/test-config.yaml Fri May 17 11:21:52 2013
@@ -1,0 +1,25 @@
+testinfo:
+    summary:     'Tests One Touch Recording with features.conf.'
+    description: |
+        'Run a SIPp scenario that tests the One Touch Recording handling on
+        Asterisk instances where automixmon is properly configured in
+        features.conf.'
+
+test-modules:
+    test-object:
+        config-section: test-object-config

[... 353 lines stripped ...]



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