[asterisk-commits] jrose: testsuite/asterisk/trunk r4096 - in /asterisk/trunk/tests/channels/SIP...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Aug 28 17:17:32 CDT 2013
Author: jrose
Date: Wed Aug 28 17:17:26 2013
New Revision: 4096
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=4096
Log:
testsuite: Fix sipp scenarios in sip_hold_direct_media to use directmedia
Humorously, for Asterisk 11 and below at least, this test would not use
directmedia due to lack of an SDP in the SIP responses. Asterisk 12 would
try to use directmedia, but the lack of SDPs caused it to behave awkwardly.
The sipp scenarios have been modified with full intention of the calls
using directmedia.
(closes issue ASTERISK-22217)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2796/
Added:
asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A_IP_restrict.xml (with props)
Removed:
asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/inject_bypass.csv
Modified:
asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/run-test
asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A.xml
asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_IP_media_restrict.xml
asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_IP_restrict.xml
asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_media_restrict.xml
Modified: asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/run-test?view=diff&rev=4096&r1=4095&r2=4096
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/run-test (original)
+++ asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/run-test Wed Aug 28 17:17:26 2013
@@ -19,8 +19,7 @@
from twisted.internet import reactor
from asterisk.version import AsteriskVersion
-logger = logging.getLogger(__name__)
-INJECT_FILE_BYPASS = "inject_bypass.csv"
+LOGGER = logging.getLogger(__name__)
INJECT_FILE_BRIDGE = "inject_bridge.csv"
class SIPHold(TestCase):
@@ -36,18 +35,16 @@
else:
self.asterisk12Events = True
- 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},
+ self.sipp_phone_a_scenarios = [
{'scenario':'phone_A.xml','-i':'127.0.0.2','-p':'5060','-inf':INJECT_FILE_BRIDGE},
+ {'scenario':'phone_A_IP_restrict.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},
+ ]
+ self.sipp_phone_b_scenarios = [
{'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},]
+ {'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
@@ -65,7 +62,7 @@
else:
ami.registerEvent('MusicOnHold', self.moh_event_handler)
- logger.info("Starting SIP scenario")
+ LOGGER.info("Starting SIP scenario")
self.execute_scenarios()
def execute_scenarios(self):
@@ -85,7 +82,7 @@
return result
if self.__test_counter == len(self.sipp_phone_a_scenarios):
- logger.info("All scenarios executed")
+ LOGGER.info("All scenarios executed")
return
sipp_a = SIPpScenario(self.test_name, self.sipp_phone_a_scenarios[self.__test_counter])
@@ -105,23 +102,23 @@
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")
+ LOGGER.info("All user events received; stopping reactor")
self.stop_reactor()
def moh_start_event_handler(self, ami, event):
- logger.debug("Received MOH start event")
+ LOGGER.debug("Received MOH start event")
self.moh_start_events += 1
def moh_stop_event_handler(self, ami, event):
- logger.debug("Received MOH stop event")
+ LOGGER.debug("Received MOH stop event")
self.moh_stop_events += 1
def moh_event_handler(self, ami, event):
if event['state'] == "Start":
- logger.debug("Received MOH start event")
+ LOGGER.debug("Received MOH start event")
self.moh_start_events += 1
elif event['state'] == "Stop":
- logger.debug("Received MOH stop event")
+ LOGGER.debug("Received MOH stop event")
self.moh_stop_events += 1
def run(self):
@@ -136,13 +133,13 @@
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))
+ 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))
+ 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))
+ 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:
Modified: asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A.xml?view=diff&rev=4096&r1=4095&r2=4096
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A.xml (original)
+++ asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A.xml Wed Aug 28 17:17:26 2013
@@ -3,6 +3,7 @@
<scenario name="Phone A Hold with IP and Media Restrictions">
+ <!-- Initial invite - Call phone B -->
<send retrans="500">
<![CDATA[
INVITE sip:[field2]@[remote_ip]:[remote_port] SIP/2.0
@@ -58,6 +59,146 @@
]]>
</send>
+ <!-- Reinvite to establish directmedia - media flows between Phone A and Phone B -->
+ <recv request="INVITE"/>
+
+ <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
+ Testsuite-Track-Phone-A: 1
+ 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 request="ACK"/>
+
+ <!-- Reinvite received for holding - media flows between Phone A and Asterisk -->
+ <recv request="INVITE"/>
+
+ <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
+ Testsuite-Track-Phone-A: 2
+ 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 request="ACK"/>
+
+ <!-- Reinvite received when unholding - media flows between Phone A and Phone B -->
+ <recv request="INVITE"/>
+
+ <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
+ Testsuite-Track-Phone-A: 3
+ 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 request="ACK"/>
+
+ <!-- Reinvite received when phone B hangs up - media flows between phone A and Asterisk -->
+ <recv request="INVITE"/>
+
+ <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
+ Testsuite-Track-Phone-A: 4
+ 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 request="ACK"/>
+
<recv request="BYE"/>
<send retrans="500">
@@ -73,9 +214,11 @@
Supported: 100rel,replaces
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
+ Testsuite-Track-Phone-A: 5
Content-Type: application/sdp
Content-Length: 0
]]>
</send>
</scenario>
+
Added: asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A_IP_restrict.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A_IP_restrict.xml?view=auto&rev=4096
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A_IP_restrict.xml (added)
+++ asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A_IP_restrict.xml Wed Aug 28 17:17:26 2013
@@ -1,0 +1,258 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone A Hold with IP and Media Restrictions">
+
+ <!-- Initial invite - Call phone B -->
+ <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>
+
+ <!-- Reinvite prior to establishing directmedia - media is set to come from 0.0.0.0 -->
+ <recv request="INVITE"/>
+
+ <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
+ Testsuite-Track-Phone-A-IP-Restrict: 1
+ 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 request="ACK"/>
+
+ <!-- Reinvite to establish directmedia - media flows between Phone A and Phone B -->
+ <recv request="INVITE"/>
+
+ <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
+ Testsuite-Track-Phone-A-IP-Restrict: 2
+ 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 request="ACK"/>
+
+ <!-- Reinvite received for holding - media flows between Phone A and Asterisk -->
+ <recv request="INVITE"/>
+
+ <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
+ Testsuite-Track-Phone-A-IP-Restrict: 3
+ 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 request="ACK"/>
+
+ <!-- Reinvite received for unholding - media flows between Phone A and Phone B -->
+ <recv request="INVITE"/>
+
+ <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
+ Testsuite-Track-Phone-A-IP-Restrict: 4
+ 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 request="ACK"/>
+
+ <!-- Reinvite receved when Phone B hangs up - media flows between Phone A and Asterisk -->
+ <recv request="INVITE"/>
+
+ <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
+ Testsuite-Track-Phone-A-IP-Restrict: 5
+ 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 request="ACK"/>
+
+ <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_direct_media/sipp/phone_A_IP_restrict.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A_IP_restrict.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_A_IP_restrict.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_IP_media_restrict.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_IP_media_restrict.xml?view=diff&rev=4096&r1=4095&r2=4096
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_IP_media_restrict.xml (original)
+++ asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_IP_media_restrict.xml Wed Aug 28 17:17:26 2013
@@ -60,12 +60,58 @@
Supported: 100rel,replaces
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
- Content-Type: application/sdp
- Content-Length: 0
+ Testsuite-Track-Phone-B-IP-Media_Restrict: 1
+ 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 ACK -->
+ <recv request="ACK"/>
+
+ <recv request="INVITE"/>
+
+ <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
+ Testsuite-Track-Phone-B-IP-Media_Restrict: 2
+ 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 request="ACK"/>
<!-- Wait some period of time -->
@@ -200,4 +246,4 @@
</send>
-</scenario>
+</scenario>
Modified: asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_IP_restrict.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_IP_restrict.xml?view=diff&rev=4096&r1=4095&r2=4096
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_IP_restrict.xml (original)
+++ asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_IP_restrict.xml Wed Aug 28 17:17:26 2013
@@ -60,12 +60,58 @@
Supported: 100rel,replaces
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
- Content-Type: application/sdp
- Content-Length: 0
+ Testuite-Track-Phone-B-IP-Restrict: 1
+ 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 ACK -->
+ <recv request="ACK"/>
+
+ <recv request="INVITE"/>
+
+ <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
+ Testuite-Track-Phone-B-IP-Restrict: 2
+ 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 request="ACK"/>
<!-- Wait some period of time -->
@@ -200,4 +246,4 @@
</send>
-</scenario>
+</scenario>
Modified: asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_media_restrict.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_media_restrict.xml?view=diff&rev=4096&r1=4095&r2=4096
==============================================================================
--- asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_media_restrict.xml (original)
+++ asterisk/trunk/tests/channels/SIP/sip_hold_direct_media/sipp/phone_B_media_restrict.xml Wed Aug 28 17:17:26 2013
@@ -60,12 +60,58 @@
Supported: 100rel,replaces
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
- Content-Type: application/sdp
- Content-Length: 0
+ Testsuite-Track-Phone-B-Media-Restrict: 1
+ 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 ACK -->
+ <recv request="ACK"/>
+
+ <recv request="INVITE"/>
+
+ <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
+ Testsuite-Track-Phone-B-Media-Restrict: 2
+ 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 request="ACK"/>
<!-- Wait some period of time -->
@@ -201,4 +247,5 @@
</send>
-</scenario>
+</scenario>
+
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