[asterisk-commits] mjordan: testsuite/asterisk/trunk r2978 - in /asterisk/trunk: lib/python/aste...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jan 3 12:45:19 CST 2012
Author: mjordan
Date: Tue Jan 3 12:45:12 2012
New Revision: 2978
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=2978
Log:
Add SIP Hold Tests
This adds a test to check for proper handling of call holding within SIP.
This includes setting the audio stream to one-way, setting the connection IP
address to 0.0.0.0, and a combination thereof.
Review: https://reviewboard.asterisk.org/r/1647
Added:
asterisk/trunk/tests/channels/SIP/sip_hold/
asterisk/trunk/tests/channels/SIP/sip_hold/configs/
asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/
asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/sip.conf (with props)
asterisk/trunk/tests/channels/SIP/sip_hold/run-test (with props)
asterisk/trunk/tests/channels/SIP/sip_hold/sipp/
asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bridge.csv (with props)
asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bypass.csv (with props)
asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A.xml (with props)
asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml (with props)
asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml (with props)
asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml (with props)
asterisk/trunk/tests/channels/SIP/sip_hold/test-config.yaml (with props)
Modified:
asterisk/trunk/lib/python/asterisk/sipp.py
asterisk/trunk/tests/channels/SIP/tests.yaml
Modified: asterisk/trunk/lib/python/asterisk/sipp.py
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/lib/python/asterisk/sipp.py?view=diff&rev=2978&r1=2977&r2=2978
==============================================================================
--- asterisk/trunk/lib/python/asterisk/sipp.py (original)
+++ asterisk/trunk/lib/python/asterisk/sipp.py Tue Jan 3 12:45:12 2012
@@ -74,10 +74,6 @@
'-i' : '127.0.0.1',
'-timeout' : '20s'
}
-
- # Override and extend defaults
- default_args.update(self.scenario)
- del default_args['scenario']
# Override and extend defaults
default_args.update(self.scenario)
Added: asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/extensions.conf?view=auto&rev=2978
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/extensions.conf Tue Jan 3 12:45:12 2012
@@ -1,0 +1,16 @@
+[general]
+PHONE_TO_DIAL=SIP/phone_B
+
+[default]
+; Dial with t/T options (force a bypass of remote and local RTP bridges),
+; even though we won't use any features for this test
+exten => bypassbridge,1,NoOp()
+ same => n,Dial(SIP/phone_B,,tTg)
+ same => n,UserEvent(TestStatus, extension: bypassbridge)
+ same => n,Hangup()
+
+; Dial with no options; use bridge set up based on peer definitions
+exten => basicdial,1,NoOp()
+ same => n,Dial(SIP/phone_B,,g)
+ same => n,UserEvent(TestStatus, extension: basicdial)
+ same => n,Hangup()
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/extensions.conf
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/extensions.conf
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/extensions.conf
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/sip.conf?view=auto&rev=2978
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/sip.conf (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/sip.conf Tue Jan 3 12:45:12 2012
@@ -1,0 +1,25 @@
+[general]
+allowguest=no
+bindaddr=127.0.0.1
+sipdebug = yes
+
+[phone_A]
+type=friend
+context=default
+disallow=all
+allow=ulaw
+qualify = no
+insecure = invite
+host = 127.0.0.2
+nat=no
+
+[phone_B]
+type=friend
+context=default
+disallow=all
+allow=ulaw
+qualify = no
+insecure = invite
+host = 127.0.0.3
+nat=no
+
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/sip.conf
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/sip.conf
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/configs/ast1/sip.conf
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/sip_hold/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/run-test?view=auto&rev=2978
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/run-test (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/run-test Tue Jan 3 12:45:12 2012
@@ -1,0 +1,126 @@
+#!/usr/bin/env python
+'''
+Copyright (C) 2011, Digium, Inc.
+Matt Jordan <mjordan at digium.com>
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+'''
+
+import sys
+import os
+import logging
+
+sys.path.append("lib/python")
+
+from asterisk.asterisk import Asterisk
+from asterisk.TestCase import TestCase
+from asterisk.sipp import SIPpScenario
+from twisted.internet import reactor
+
+logger = logging.getLogger(__name__)
+TEST_DIR = os.path.dirname(os.path.realpath(__file__))
+INJECT_FILE_BYPASS = TEST_DIR + "/sipp/inject_bypass.csv"
+INJECT_FILE_BRIDGE = TEST_DIR + "/sipp/inject_bridge.csv"
+
+class SIPHold(TestCase):
+ def __init__(self):
+ TestCase.__init__(self)
+ self.create_asterisk()
+ self.sipp_phone_a_scenarios = [{'scenario':'phone_A.xml','-i':'127.0.0.2','-p':'5060','-inf':INJECT_FILE_BYPASS},
+ {'scenario':'phone_A.xml','-i':'127.0.0.2','-p':'5060','-inf':INJECT_FILE_BYPASS},
+ {'scenario':'phone_A.xml','-i':'127.0.0.2','-p':'5060','-inf':INJECT_FILE_BYPASS},
+ {'scenario':'phone_A.xml','-i':'127.0.0.2','-p':'5060','-inf':INJECT_FILE_BRIDGE},
+ {'scenario':'phone_A.xml','-i':'127.0.0.2','-p':'5060','-inf':INJECT_FILE_BRIDGE},
+ {'scenario':'phone_A.xml','-i':'127.0.0.2','-p':'5060','-inf':INJECT_FILE_BRIDGE},]
+ self.sipp_phone_b_scenarios = [{'scenario':'phone_B_media_restrict.xml','-i':'127.0.0.3','-p':'5060','-inf':INJECT_FILE_BYPASS},
+ {'scenario':'phone_B_IP_restrict.xml','-i':'127.0.0.3','-p':'5060','-inf':INJECT_FILE_BYPASS},
+ {'scenario':'phone_B_IP_media_restrict.xml','-i':'127.0.0.3','-p':'5060','-inf':INJECT_FILE_BYPASS},
+ {'scenario':'phone_B_media_restrict.xml','-i':'127.0.0.3','-p':'5060','-inf':INJECT_FILE_BRIDGE},
+ {'scenario':'phone_B_IP_restrict.xml','-i':'127.0.0.3','-p':'5060','-inf':INJECT_FILE_BRIDGE},
+ {'scenario':'phone_B_IP_media_restrict.xml','-i':'127.0.0.3','-p':'5060','-inf':INJECT_FILE_BRIDGE},]
+
+ self.passed = True
+ self.moh_start_events = 0
+ self.moh_stop_events = 0
+ self.user_events = 0
+
+ def ami_connect(self, ami):
+ TestCase.ami_connect(self, ami)
+ ami.registerEvent('UserEvent', self.user_event_handler)
+ ami.registerEvent('MusicOnHold', self.moh_event_handler)
+ logger.info("Starting SIP scenario")
+ self.execute_scenarios()
+
+ def execute_scenarios(self):
+ for i in range(len(self.sipp_phone_a_scenarios)):
+ sipp_a = SIPpScenario(TEST_DIR, self.sipp_phone_a_scenarios[i])
+ sipp_b = SIPpScenario(TEST_DIR, self.sipp_phone_b_scenarios[i])
+
+ """ Start up the listener first - Phone A calls Phone B """
+ sipp_b.run()
+ sipp_a.run()
+
+ sipp_a_result = sipp_a.waitAndEvaluate()
+ sipp_b_result = sipp_b.waitAndEvaluate()
+
+ if (not sipp_a_result):
+ logger.warn("SIPp Scenario Phone A (%s) failed" % self.sipp_phone_a_scenarios[i]['scenario'])
+ self.passed = False
+ if (not sipp_b_result):
+ logger.warn("SIPp Scenario Phone B (%s) failed" % self.sipp_phone_b_scenarios[i]['scenario'])
+ self.passed = False
+ self.reset_timeout()
+
+ logger.info("All scenarios executed")
+ """
+ Note: you can't stop the reactor here, as the AMI events will be pooled up. Let the AMI events
+ determine when the reactor is stopped
+ """
+
+ def user_event_handler(self, ami, event):
+ self.user_events += 1
+ if (self.user_events == len(self.sipp_phone_a_scenarios)):
+ logger.info("All user events received; stopping reactor")
+ self.stop_reactor()
+
+ def moh_event_handler(self, ami, event):
+ if event['state'] == "Start":
+ logger.debug("Received MOH start event")
+ self.moh_start_events += 1
+ elif event['state'] == "Stop":
+ logger.debug("Received MOH stop event")
+ self.moh_stop_events += 1
+
+ def run(self):
+ TestCase.run(self)
+ self.create_ami_factory()
+
+
+def main():
+ test = SIPHold()
+ test.start_asterisk()
+ reactor.run()
+ test.stop_asterisk()
+
+ if (test.moh_start_events != len(test.sipp_phone_a_scenarios)):
+ logger.error("Failed to receive %d MOH start events (received %d)" % (len(test.sipp_phone_a_scenarios), test.moh_start_events))
+ test.passed = False
+ if (test.moh_stop_events != len(test.sipp_phone_a_scenarios)):
+ logger.error("Failed to receive %d MOH stop events (received %d)" % (len(test.sipp_phone_a_scenarios), test.moh_stop_events))
+ test.passed = False
+ if (test.user_events != len(test.sipp_phone_a_scenarios)):
+ logger.error("Failed to receive %d user test events (received %d)" % (len(test.sipp_phone_a_scenarios), test.user_events))
+ test.passed = False
+
+ if test.passed:
+ return 0
+ else:
+ return 1
+
+
+if __name__ == "__main__":
+ sys.exit(main())
+
+
+# vim:sw=4:ts=4:expandtab:textwidth=79
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/run-test
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/run-test
------------------------------------------------------------------------------
svn:executable = *
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/run-test
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/run-test
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bridge.csv
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bridge.csv?view=auto&rev=2978
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bridge.csv (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bridge.csv Tue Jan 3 12:45:12 2012
@@ -1,0 +1,2 @@
+SEQUENTIAL
+phone_A;phone_B;basicdial
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bridge.csv
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bridge.csv
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bridge.csv
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bypass.csv
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bypass.csv?view=auto&rev=2978
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bypass.csv (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bypass.csv Tue Jan 3 12:45:12 2012
@@ -1,0 +1,2 @@
+SEQUENTIAL
+phone_A;phone_B;bypassbridge
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bypass.csv
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bypass.csv
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/inject_bypass.csv
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A.xml?view=auto&rev=2978
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A.xml (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A.xml Tue Jan 3 12:45:12 2012
@@ -1,0 +1,81 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone A Hold with IP and Media Restrictions">
+
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:[field2]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
+ To: <sip:[field2]@[remote_ip]:[remote_port];user=phone>
+ CSeq: 1 INVITE
+ Call-ID: [call_id]
+ Contact: <sip:[field0]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Allow-Events: talk,hold,conference
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ a=sendrecv
+ 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="183" optional="true" />
+
+ <recv response="200" />
+
+ <send>
+ <![CDATA[
+ ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field0] <sip:[field0]@[remote_ip]>;tag=[call_number]
+ To: <sip:[field1]@[remote_ip];user=phone>[peer_tag_param]
+ CSeq: 1 ACK
+ Call-ID: [call_id]
+ Contact: <sip:[field0]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv request="BYE"/>
+
+ <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:[field0]@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ Supported: 100rel,replaces
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Type: application/sdp
+ Content-Length: 0
+ ]]>
+ </send>
+
+</scenario>
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml?view=auto&rev=2978
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml Tue Jan 3 12:45:12 2012
@@ -1,0 +1,203 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone B Hold with IP and Media Restrictions">
+ <Global variables="global_call_id"/>
+
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp=".*"
+ header="Call-ID:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="global_call_id"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Allow-Events: talk,hold,conference
+ Accept-Language: en
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <pause milliseconds="200"/>
+
+ <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:[field1]@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ Supported: 100rel,replaces
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Type: application/sdp
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- RECV ACK -->
+ <recv request="ACK"/>
+
+ <!-- Wait some period of time -->
+ <pause milliseconds="3000"/>
+
+ <!-- Modify RTP session to be send only -->
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+ To: [field0] <sip:[field1]@[remote_ip]>
+ CSeq: [cseq] INVITE
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Supported: 100rel,replaces
+ Allow-Events: talk,hold,conference
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1325003603 1325003604 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 0.0.0.0
+ t=0 0
+ m=audio 2226 RTP/AVP 0 101
+ a=sendonly
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ ]]>
+ </send>
+
+ <recv response="100" optional="true" />
+
+ <recv response="200" />
+
+ <pause milliseconds="200"/>
+
+ <send>
+ <![CDATA[
+ ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
+ To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Wait some period of time, then send the un-hold -->
+ <pause milliseconds="3000"/>
+
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+ To: [field0] <sip:[field1]@[remote_ip]>
+ CSeq: [cseq] INVITE
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Supported: 100rel,replaces
+ Allow-Events: talk,hold,conference
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1325003603 1325003605 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ a=sendrecv
+ m=audio 2226 RTP/AVP 0 101
+ a=sendrecv
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ ]]>
+ </send>
+
+ <recv response="100" optional="true" />
+
+ <recv response="200" />
+
+ <send>
+ <![CDATA[
+ ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
+ To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Wait some period of time -->
+ <pause milliseconds="500"/>
+
+ <send>
+ <![CDATA[
+ BYE sip:[field1]@1[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+ To: [field0] <sip:[field1]@[remote_ip]>[peer_tag_param]
+ CSeq: [cseq] BYE
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+
+</scenario>
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml?view=auto&rev=2978
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml Tue Jan 3 12:45:12 2012
@@ -1,0 +1,203 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone B Hold with IP and Media Restrictions">
+ <Global variables="global_call_id"/>
+
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp=".*"
+ header="Call-ID:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="global_call_id"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Allow-Events: talk,hold,conference
+ Accept-Language: en
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <pause milliseconds="200"/>
+
+ <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:[field1]@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ Supported: 100rel,replaces
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Type: application/sdp
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- RECV ACK -->
+ <recv request="ACK"/>
+
+ <!-- Wait some period of time -->
+ <pause milliseconds="3000"/>
+
+ <!-- Modify RTP session to be send only -->
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+ To: [field0] <sip:[field1]@[remote_ip]>
+ CSeq: [cseq] INVITE
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Supported: 100rel,replaces
+ Allow-Events: talk,hold,conference
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1325003603 1325003604 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 0.0.0.0
+ 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="200" />
+
+ <pause milliseconds="200"/>
+
+ <send>
+ <![CDATA[
+ ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
+ To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Wait some period of time, then send the un-hold -->
+ <pause milliseconds="3000"/>
+
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+ To: [field0] <sip:[field1]@[remote_ip]>
+ CSeq: [cseq] INVITE
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Supported: 100rel,replaces
+ Allow-Events: talk,hold,conference
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1325003603 1325003605 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ a=sendrecv
+ 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="200" />
+
+ <send>
+ <![CDATA[
+ ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
+ To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Wait some period of time -->
+ <pause milliseconds="500"/>
+
+ <send>
+ <![CDATA[
+ BYE sip:[field1]@1[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+ To: [field0] <sip:[field1]@[remote_ip]>[peer_tag_param]
+ CSeq: [cseq] BYE
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+
+</scenario>
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml?view=auto&rev=2978
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml Tue Jan 3 12:45:12 2012
@@ -1,0 +1,204 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone B Hold with Media Restrictions">
+ <Global variables="global_call_id"/>
+
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp=".*"
+ header="Call-ID:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="global_call_id"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Allow-Events: talk,hold,conference
+ Accept-Language: en
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <pause milliseconds="200"/>
+
+ <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:[field1]@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ Supported: 100rel,replaces
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Type: application/sdp
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- RECV ACK -->
+ <recv request="ACK"/>
+
+ <!-- Wait some period of time -->
+ <pause milliseconds="3000"/>
+
+ <!-- Modify RTP session to be send only -->
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+ To: [field0] <sip:[field1]@[remote_ip]>
+ CSeq: [cseq] INVITE
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Supported: 100rel,replaces
+ Allow-Events: talk,hold,conference
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1325003603 1325003604 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ a=sendonly
+ m=audio 2226 RTP/AVP 0 101
+ a=sendonly
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ ]]>
+ </send>
+
+ <recv response="100" optional="true" />
+
+ <recv response="200" />
+
+ <pause milliseconds="200"/>
+
+ <send>
+ <![CDATA[
+ ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
+ To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Wait some period of time, then send the un-hold -->
+ <pause milliseconds="3000"/>
+
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+ To: [field0] <sip:[field1]@[remote_ip]>
+ CSeq: [cseq] INVITE
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Supported: 100rel,replaces
+ Allow-Events: talk,hold,conference
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1325003603 1325003605 IN IP4 [local_ip]
+ s=Polycom IP Phone
+ c=IN IP4 [local_ip]
+ t=0 0
+ a=sendrecv
+ m=audio 2226 RTP/AVP 0 101
+ a=sendrecv
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ ]]>
+ </send>
+
+ <recv response="100" optional="true" />
+
+ <recv response="200" />
+
+ <send>
+ <![CDATA[
+ ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
+ To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <!-- Wait some period of time -->
+ <pause milliseconds="500"/>
+
+ <send>
+ <![CDATA[
+ BYE sip:[field1]@1[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch]
+ From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
+ To: [field0] <sip:[field1]@[remote_ip]>[peer_tag_param]
+ CSeq: [cseq] BYE
+ Call-ID: [$global_call_id]
+ Contact: <sip:[field1]@[local_ip]:[local_port]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+
+</scenario>
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/sip_hold/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold/test-config.yaml?view=auto&rev=2978
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold/test-config.yaml Tue Jan 3 12:45:12 2012
@@ -1,0 +1,17 @@
+testinfo:
+ summary: 'Test various SIP Hold scenarios'
+ description: |
+ This tests SIP Hold, where one SIP phone puts another SIP phone on hold by
+ sending a re-INVITE with a modified SDP containing either a restricted audio
+ direction, an IP address of 0.0.0.0, or a combination thereof. This
+ is tested both for a local RTP bridge, and a non-bridged scenario.
+
+properties:
+ minversion: '1.8.9'
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ testconditions:
+ - name: 'threads'
+ ignoredThreads:
+ - 'autoservice_run'
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/test-config.yaml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/test-config.yaml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold/test-config.yaml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: asterisk/trunk/tests/channels/SIP/tests.yaml
[... 10 lines stripped ...]
More information about the asterisk-commits
mailing list