[Asterisk-code-review] A testsuite test for 3PCC patch for AMI "SIPnotify" (ASTERIS... (testsuite[master])

Yasuhiko Kamata asteriskteam at digium.com
Thu Dec 21 00:54:10 CST 2017


Yasuhiko Kamata has uploaded this change for review. ( https://gerrit.asterisk.org/7705


Change subject: A testsuite test for 3PCC patch for AMI "SIPnotify" (ASTERISK-27461).
......................................................................

A testsuite test for 3PCC patch for AMI "SIPnotify" (ASTERISK-27461).

This test is somewhat complicated than other tests in AMI;
because the value of "Call-ID" is needed to send "SIPnotify" AMI action.

Change-Id: Idccbf32ed6670a5205ee99bd7413c7fe0804efb1
---
A tests/channels/SIP/ami/sip_notify/call_id/configs/ast1/extensions.conf
A tests/channels/SIP/ami/sip_notify/call_id/configs/ast1/manager.conf
A tests/channels/SIP/ami/sip_notify/call_id/configs/ast1/sip.conf
A tests/channels/SIP/ami/sip_notify/call_id/run-test
A tests/channels/SIP/ami/sip_notify/call_id/sipp/callee.xml
A tests/channels/SIP/ami/sip_notify/call_id/sipp/caller.xml
A tests/channels/SIP/ami/sip_notify/call_id/test-config.yaml
M tests/channels/SIP/ami/sip_notify/tests.yaml
8 files changed, 360 insertions(+), 0 deletions(-)



  git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/05/7705/1

diff --git a/tests/channels/SIP/ami/sip_notify/call_id/configs/ast1/extensions.conf b/tests/channels/SIP/ami/sip_notify/call_id/configs/ast1/extensions.conf
new file mode 100644
index 0000000..106b4ff
--- /dev/null
+++ b/tests/channels/SIP/ami/sip_notify/call_id/configs/ast1/extensions.conf
@@ -0,0 +1,3 @@
+[default]
+exten => callee,1,Dial(SIP/callee)
+
diff --git a/tests/channels/SIP/ami/sip_notify/call_id/configs/ast1/manager.conf b/tests/channels/SIP/ami/sip_notify/call_id/configs/ast1/manager.conf
new file mode 100644
index 0000000..c6deda4
--- /dev/null
+++ b/tests/channels/SIP/ami/sip_notify/call_id/configs/ast1/manager.conf
@@ -0,0 +1,10 @@
+[general]
+enabled = yes
+port = 5038
+
+[user]
+secret = mysecret
+permit = 127.0.0.1/255.255.255.255
+read = all
+write = all
+
diff --git a/tests/channels/SIP/ami/sip_notify/call_id/configs/ast1/sip.conf b/tests/channels/SIP/ami/sip_notify/call_id/configs/ast1/sip.conf
new file mode 100644
index 0000000..d782b39
--- /dev/null
+++ b/tests/channels/SIP/ami/sip_notify/call_id/configs/ast1/sip.conf
@@ -0,0 +1,21 @@
+[general]
+udpbindaddr=0.0.0.0:5060
+
+[caller]
+type=friend
+host=127.0.0.1
+port=5062
+directmedia=no
+disallow=all
+allow=ulaw
+context=default
+
+[callee]
+type=friend
+host=127.0.0.1
+port=5063
+directmedia=no
+disallow=all
+allow=ulaw
+context=default
+
diff --git a/tests/channels/SIP/ami/sip_notify/call_id/run-test b/tests/channels/SIP/ami/sip_notify/call_id/run-test
new file mode 100755
index 0000000..12e8d82
--- /dev/null
+++ b/tests/channels/SIP/ami/sip_notify/call_id/run-test
@@ -0,0 +1,199 @@
+#!/usr/bin/env python
+
+import sys
+import os
+import logging
+import time
+import socket
+import threading
+import errno
+
+sys.path.append("lib/python")
+sys.path.append("utils")
+
+from tempfile import NamedTemporaryFile
+from twisted.internet import reactor
+from asterisk.sipp import SIPpTest
+
+WORKING_DIR = os.path.abspath(os.path.dirname(__file__))
+TEST_DIR = os.path.dirname(os.path.realpath(__file__))
+
+logger = logging.getLogger(__name__)
+
+class DupDict(dict):
+    def __setitem__(self, key, value):
+        try:
+            self[key]
+        except KeyError:
+            super(DupDict, self).__setitem__(key, [])
+        self[key].append(value)
+
+class AmiTestClientThread(threading.Thread):
+    """ AMI test client thread """
+
+    def __init__(self):
+        super(AmiTestClientThread, self).__init__()
+
+    def set_amihost(self, host, port):
+        self.host = host
+        self.port = port
+
+    def connect(self):
+        self.ami_client = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
+
+        while True:
+            try:
+                self.ami_client.connect((self.host, self.port))
+                break
+            except socket.error as serr:
+                if serr.errno != errno.ECONNREFUSED:
+                    raise serr
+                time.sleep(0.5) # retry wait
+
+        self.ami_client_file = self.ami_client.makefile()
+
+    def close(self):
+        self.ami_client_file.close()
+        self.ami_client.close()
+
+    def receive_message(self):
+        message = DupDict()
+
+        while True:
+            linedata = self.ami_client_file.readline()
+
+            linedata = linedata.replace("\r", "")
+            linedata = linedata.replace("\n", "")
+
+            if len(linedata) < 1:
+                break
+
+            keyvalues = linedata.split(":", 1)
+            if len(keyvalues) > 1:
+                message[keyvalues[0].strip()] = keyvalues[1].strip()
+            else:
+                message[linedata.strip()] = ""
+
+        return message
+
+    def send_message(self, message):
+        messagetext = ""
+
+        for key, value in message.iteritems():
+            messagetext += key + ": " + value + "\r\n"
+        messagetext += "\r\n"
+
+        self.ami_client.send(messagetext)
+
+    def run(self):
+        # connect and receive banner
+        self.connect()
+        self.banner = self.ami_client_file.readline()
+
+        # login
+        message = { "Action": "Login", "Username": "user", "Secret": "mysecret" }
+        self.send_message(message)
+
+        # wait for "Response: Success"
+        while True:
+            message = self.receive_message()
+            if message.has_key("Response"):
+                if message["Response"][0] == "Success":
+                    break
+                else:
+                    print "Response is " + str(message["Response"])
+                    raise
+
+        # wait for "Event: Newchannel", "Channel: SIP/callee..."
+        while True:
+            message = self.receive_message()
+            if message.has_key("Event"):
+                if message["Event"][0] == "Newchannel":
+                    if message["Channel"][0][0:10] == "SIP/callee":
+                        break
+        channel = message["Channel"][0]
+
+        # send "Action: Status"
+        message = { "Action": "Status", "Channel": channel, "AllVariables": "true" }
+        self.send_message(message)
+
+        # wait for "Event: Status"
+        call_id = ""
+        while True:
+            message = self.receive_message()
+            if message.has_key("Event"):
+                if message["Event"][0] == "Status":
+                    if message.has_key("Variable"):
+                        for value in message["Variable"]:
+                            if value[0:10] == "SIPCALLID=":
+                              call_id = value[10:]
+                    break
+        if len(call_id) < 1:
+            raise
+
+        # send "Action: SIPnotify"
+        message = { "Action": "SIPnotify", "Channel": channel, "Variable": "Event=talk", "Call-ID": call_id }
+        self.send_message(message)
+
+        # wait for "Response: Success"
+        while True:
+            message = self.receive_message()
+            if message.has_key("Response"):
+                if message["Response"][0] == "Success":
+                    break
+                else:
+                    print "Response is " + str(message["Response"])
+                    raise
+
+        self.close()
+
+def main():
+    sipplog = NamedTemporaryFile(delete=True)
+
+    SIPP_SCENARIOS = [
+        {
+            'scenario': 'caller.xml',
+            '-i': '127.0.0.1',
+            '-p': '5062',
+            '-s': '3200000000',
+            '-message_file': '/tmp/caller.log', # sipplog.name,
+            # Cheat and pass two argumentless options as key and value
+            # because the SIPpTest doesn't allow us to pass ordered-args.
+            # We use -pause_msg_ign to ignore messages while being paused
+            # and then check the log (from -trace_msg) for those messages.
+            '-trace_msg': '-pause_msg_ign',
+        },
+        {
+            'scenario': 'callee.xml',
+            '-i': '127.0.0.1',
+            '-p': '5063',
+            '-s': '3200000000',
+            '-message_file': '/tmp/callee.log', # sipplog.name,
+            # Cheat and pass two argumentless options as key and value
+            # because the SIPpTest doesn't allow us to pass ordered-args.
+            # We use -pause_msg_ign to ignore messages while being paused
+            # and then check the log (from -trace_msg) for those messages.
+            '-trace_msg': '-pause_msg_ign',
+        }
+    ]
+
+    test = SIPpTest(WORKING_DIR, TEST_DIR, SIPP_SCENARIOS)
+    test.reactor_timeout = 10
+
+    ami_client = AmiTestClientThread()
+    ami_client.set_amihost("localhost", 5038)
+    ami_client.start()
+
+    reactor.run()
+
+    # If it failed, bail.
+    if not test.passed:
+        return 1
+
+    return 0
+
+
+if __name__ == "__main__":
+    sys.exit(main())
+
+# vim:sw=4:ts=4:expandtab:textwidth=79
diff --git a/tests/channels/SIP/ami/sip_notify/call_id/sipp/callee.xml b/tests/channels/SIP/ami/sip_notify/call_id/sipp/callee.xml
new file mode 100755
index 0000000..d3bff3c
--- /dev/null
+++ b/tests/channels/SIP/ami/sip_notify/call_id/sipp/callee.xml
@@ -0,0 +1,76 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Notify Request with Call-ID">
+
+    <recv request="INVITE">
+        <action>
+            <ereg regexp=": .*"
+                search_in="hdr"
+                header="Call-ID"
+                check_it="true"
+                assign_to="1"/>
+            <ereg regexp=": .*"
+                search_in="hdr"
+                header="CSeq"
+                check_it="true"
+                assign_to="2"/>
+            <log message="Received INVITE with Call-ID [$1] and CSeq [$2]." />
+        </action>
+    </recv>
+
+    <send>
+      <![CDATA[
+
+      SIP/2.0 180 Ringing
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag[call_number]
+      Call-ID: [call_id]
+      [last_CSeq:]
+      Contact: <sip:user1@[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+      ]]>
+    </send>
+
+    <recv request="NOTIFY">
+        <action>
+            <ereg regexp=": .*$"
+                search_in="hdr"
+                header="Call-ID"
+                check_it="true"
+                assign_to="3"/>
+            <log message="Received NOTIFY with Call-ID [$3]." />
+        </action>
+    </recv>
+
+    <send>
+      <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag[call_number]
+      Call-ID: [call_id]
+      [last_CSeq:]
+      Contact: <sip:user1@[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]SIPpTag[call_number]
+      Call-ID: [call_id]
+      CSeq[$2]
+      Contact: <sip:user1@[local_ip]:[local_port];transport=[transport]>
+      Content-Type: application/sdp
+      Content-Length: [len]
+      ]]>
+    </send>
+
+</scenario>
diff --git a/tests/channels/SIP/ami/sip_notify/call_id/sipp/caller.xml b/tests/channels/SIP/ami/sip_notify/call_id/sipp/caller.xml
new file mode 100644
index 0000000..373d361
--- /dev/null
+++ b/tests/channels/SIP/ami/sip_notify/call_id/sipp/caller.xml
@@ -0,0 +1,35 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Notify Request with Call-ID">
+
+  <send retrans="500">
+    <![CDATA[
+      INVITE sip:callee at voxbone.com SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller at voxbone.com>;tag=[call_number]
+      To: callee <sip:callee at voxbone.com:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Content-Type: application/sdp
+      Content-Length: [len]
+      v=0
+      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[local_ip_type] [local_ip]
+      t=0 0
+      m=audio 9000 RTP/AVP 8
+      a=rtpmap:8 PCMU/8000
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="200">
+  </recv>
+</scenario>
diff --git a/tests/channels/SIP/ami/sip_notify/call_id/test-config.yaml b/tests/channels/SIP/ami/sip_notify/call_id/test-config.yaml
new file mode 100644
index 0000000..47c9199
--- /dev/null
+++ b/tests/channels/SIP/ami/sip_notify/call_id/test-config.yaml
@@ -0,0 +1,15 @@
+info:
+    summary: 'Test SIPNotify AMI Action for Call-ID'
+    description: |
+        This Tests the AMI Action SIPNotify in order to make sure
+        that Call-ID header can be specified.
+
+properties:
+    minversion: '12.0.0'
+    dependencies:
+        - sipp :
+            version : 'v3.0'
+        - asterisk : 'chan_sip'
+    tags:
+        - SIP
+
diff --git a/tests/channels/SIP/ami/sip_notify/tests.yaml b/tests/channels/SIP/ami/sip_notify/tests.yaml
index b1a3008..750f6d6 100644
--- a/tests/channels/SIP/ami/sip_notify/tests.yaml
+++ b/tests/channels/SIP/ami/sip_notify/tests.yaml
@@ -2,3 +2,4 @@
 tests:
     - test: 'custom_headers'
     - test: 'content'
+    - test: 'call_id'

-- 
To view, visit https://gerrit.asterisk.org/7705
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-MessageType: newchange
Gerrit-Change-Id: Idccbf32ed6670a5205ee99bd7413c7fe0804efb1
Gerrit-Change-Number: 7705
Gerrit-PatchSet: 1
Gerrit-Owner: Yasuhiko Kamata <yasuhiko.kamata at nxtg.co.jp>
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