Prerelease Phone Feature Test
Content
Pre-Release Phone Feature Test
(This test must be run at least once for each release.)
Test Date:
Tester:
Phones Tested:
Platforms:
Phone Test version:
Final Version No:
Particular attention was focused on phone features
This version is for use on Teo UCM, Avaya, Redcom, BroadSoft, Cisco, NEC
TABLE OF CONTENTS
CONTENT
1
2
3
4
5
6
7
8
8.1
8.2
8.3
8.3.1
8.3.2
8.3.3
8.3.4
8.4
8.4.1
8.4.2
8.4.3
8.5
8.6
8.6.1
8.6.2
8.7
8.7.1
8.7.2
8.8
8.8.1
8.8.2
8.8.3
PAGE NUMBER
Scope.................................................................................................................................................... 2
Hardware used for the test.................................................................................................................... 2
Test Servers.......................................................................................................................................... 2
Software versions used during the test................................................................................................. 2
Test Responses..................................................................................................................................... 2
Test Failures.......................................................................................................................................... 3
Order of Test......................................................................................................................................... 3
Regression Tests................................................................................................................................... 3
INSTALL: RESET, MAC xml.................................................................................................................. 3
INSTALL: RESET, LINE xml.................................................................................................................. 4
PROGRAM UPDATE............................................................................................................................ 4
INSTALL: UPDATE, PRGRM, TFTP Program Update..........................................................................4
INSTALL: UPDATE, PRGRM, Program Update Not Required..............................................................5
INSTALL: UPDATE, PRGRM, HTTP or HTTPS Program Update.........................................................5
INSTALL: UPDATE, PRGRM, TEO....................................................................................................... 6
USER: VOICE, VOLUME...................................................................................................................... 7
USER: VOICE, VOLUME, Receive....................................................................................................... 7
USER: VOICE, VOLUME, Transmit...................................................................................................... 7
USER: VOICE, VOLUME, Reset to Defaults........................................................................................ 7
INSTALL: QOS, Quality of Service........................................................................................................ 7
CODEC TESTING................................................................................................................................. 9
INSTALL: Keys, Edit Line Key, CODEC................................................................................................ 9
AUTO CODEC SELECTION: Install, Keys, Edit Line Key...................................................................11
JITTER Test......................................................................................................................................... 12
JITTER, FIXED: INSTALL, KEYS, Edit Line Key................................................................................12
JITTER, ADAPTIVE: INSTALL KEYS, Edit Line Key...........................................................................13
USER: RING, Personal Ringing.......................................................................................................... 13
USER: RING, Personal Ringing, TONE.............................................................................................. 13
OFFHOOK Ringing............................................................................................................................. 13
RING CONTROL................................................................................................................................ 14
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pg. 1
8.8.4
8.8.5
8.8.6
8.9
8.9.1
8.10
8.11
8.12
8.13
8.14
8.15
8.16
9
Always Ring........................................................................................................................................ 14
Never Ring.......................................................................................................................................... 14
Ring after a Delay............................................................................................................................... 14
Adjusting the Ringer Volume............................................................................................................... 15
Distinctive Ringing of SIP Phone........................................................................................................ 15
USER: DIR, User Directory................................................................................................................. 16
Call Log............................................................................................................................................... 17
HOT/WARM Dialing............................................................................................................................ 18
ADMIN: INSPCT, Inspect Keys.......................................................................................................... 18
INSTALL: PASSWD, Installation Password.........................................................................................18
USER: PIN, Setting a Call Log PIN..................................................................................................... 18
INSTALL: IP, IP Addresses, Syslog..................................................................................................... 19
ISSUES:.............................................................................................................................................. 20
1 Scope
An engineering regression test to test common phone functions, including call control.
Testing is Ad-Hoc unless specific test steps are listed. In the case of tests with well developed steps,
there is still Ad-Hoc testing performed.
2 Hardware used for the test
7810
7810-TSG
7810 + 8030X
4104
4101
3 Test Servers
Teo UCM
4 Software versions used during the test
Phone Software: v0x.04.20
5 Test Responses
√, P: Pass
Х, F: Fail
N/T: Not Tested
N/A: Not Applicable
BUG: Issues that occurred during the test.
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pg. 2
6 Test Failures
Test Failures and reported issues are documented as Bugs in the JIRA Database, along with bug repair
and retest results. Test failures are repaired in following software versions that may or may not be part of
this final release. Bug repair priorities are determined by engineering management. To review open bugs
with lower priorities, review the JIRA Database. A basic understanding of Teo VOIP phone operation is
necessary in order to complete this test, and a focus on ad-hoc phone testing (unusual actions with a
specific focus) is also desirable to find additional problems. Tests are added as new scenarios are
discovered.
Test as appropriate on 7810, 7810 + 8030X, 7810-TSG, 4104, 4101.
7 Order of Test
This list is provided as a logical progression of events and tests to be performed.
Test
Test New Features or Bug Fixes
Program Update from the new version
to something else
Handset and Speaker volume
Voice quality test
QoS Test
Codec and P time test
Ring Controls and ringer Volume
Dialing from the Directory and Call
Log
Hot/Warm Dialing
Admin Inspect Keys
Install PIN
User PIN
Syslog Test
Time Estimate
PASS/FAIL
8 Regression Tests
8.1 INSTALL: RESET, MAC xml
Test on 7810, 7810 + 8030X, 4104, 4101
Step
1
2
Action
Configure the TCS7000A.xml
file (global xml tag file) with a
configuration tag set for MAC
address.
Expected Result
Create a unique configuration
xml file for the test phone,
with the file name made up of
PreReleasePhoneFeatureTest.doc
8/28/2014
Comments
Set the TCS7000A config tag
to MAC:
<config_file_name_base>MA
C</config_file_name_base>
Or comment out the line;
MAC is the default
i.e. 00048DFFFFE8.xml
P/F
pg. 3
Step
Action
the phone MAC address.
Reset the phone, and watch
the TFTP server window
3
Expected Result
The phone will load the
TCS7000A.xml file and its
MAC.xml file.
The phone will initialize and
register.
Comments
P/F
As the phone initializes, it will
prompt the user for needed
data.
4
8.2 INSTALL: RESET, LINE xml
Test on 7810, 7810 + 8030X, 4104, 4101
Step
1
2
Action
Configure the TCS7000A.xml
file (global xml tag file) with a
configuration tag set for LINE
address.
Create a unique configuration
xml file for the test phone,
with the file name made up of
the phone Line ID.
Reset the phone, and watch
the TFTP server window.
3
Expected Result
The phone will load the
TCS7000A.xml file and its
Line ID.xml file.
The phone will initialize and
register.
Comments
Set the TCS7000A config tag
to LINE:
<config_file_name_base>LIN
E</config_file_name_base>
P/F
As the phone initializes, it will
prompt the user for needed
data.
4
8.3 PROGRAM UPDATE
8.3.1
Step
1
2
3
4
INSTALL: UPDATE, PRGRM, TFTP Program Update
Test on 7810, 7810 + 8030X, 4104, 4101
Action
Expected Result
Provision the Update Server
with a different version of
software from the current
version running in the UUT.
Configure the UUT for TFTP
update
Comments
P/F
SETUP / INSTL / (right arrow) /
UPDATE / PRGRM / PROTO /
TFTP
SETUP / INSTL / (right arrow) /
UPDATE / PRGRM / START
Select "UPDATE / PRGRM"to The UUT should load the
load the different software
different software, and
from the Update Server.
RESTART. After RESTART the
phone should successfully
register, and have the same
LINE ID and other settings as
before the update.
Check the error log for
There should be no unexpected SETUP / INSTL / (right arrow) /
problems.
errors in the log.
(right arrow) / LOG / ERROR
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pg. 4
Step
5
Action
Expected Result
Check the phone S/W version It should be the new version.
to see that it matches the
version just loaded
Comments
P/F
SETUP / ADMIN / VERS / S/W
6
8.3.2
INSTALL: UPDATE, PRGRM, Program Update Not Required
Test on 7810, 7810 + 8030X, 4104, 4101
Step
Action
Expected Result
Comments
Provision the Update Server
SETUP / ADMIN / VERS / S/W
with the same version of
1
software that is presently
running in the phone.
Configure the UUT for TFTP
SETUP / INSTL / (right arrow) /
2
update
UPDATE / PRGRM / PROTO /
TFTP
Perform a Update Program to The UUT should not load the
SETUP / INSTL / (right arrow) /
attempt to load the same
software into the phone. The UPDATE / PRGRM / START
3
software from the Update
phone will display:
Server.
PROGRAM UPDATE NOT
REQUIRED
Check the error log for
There should be no unexpected SETUP / INSTL / (right arrow) /
4
problems.
errors in the log.
(right arrow) / LOG / ERROR
5
INSTALL: UPDATE, PRGRM, HTTP or HTTPS Program Update
Test on 7810, 7810 + 8030X, 4104, 4101
Step
Action
Expected Result
Comments
Create a location in the
Server name = phonemgrcomputer Update server path,
w2k8.greyhawk.tonecommande
for program update files. This
r.com
will be a different location
Server name with file location
than for TFTP updates.
appended = phonemgrThis path is where the
w2k8.greyhawk.tonecommande
digitally signed files are
r.com/teotestdata/stever.
located under the update
This is the location that the
server root directory.
HTTPS server is pointed at.
Place the folder containing
1
the phone update program
If you are connected to a TEO
files in the HTTP/HTTPS
UCM server, Program Updates
update file address. Also
are done with HTTPS protocol.
place all of the folder contents
This is adequate for this test.
in the same location.
P/F
8.3.3
2
Or do a Program Update from
the keypad on a phone
connected to a TEO UCM
server.
Edit the phone's UPDATE IP
address (Domain Name),
appending the location for the
PreReleasePhoneFeatureTest.doc
P/F
The program update file
location that was just created.
8/28/2014
pg. 5
Step
3
4
5
6
7
Action
program Update files:
On the phone select SETUP /
INSTL / IP / (right arrow) /
UPDATE / EDIT / DHCP4 /
OK.
Use the navigation arrows to
move the cursor to the end of
the displayed Update Server
Address. Append the
HTTP/HTTPS file location.
Expected Result
Enter the following key
sequence on the phone to set
the phone’s update protocol
to HTTP or HTTPS:
On the phone, select
SETUP / INSTL / (right arrow)
/ UPDATE / PRGRM /
PROTO / HTTP(or HTTPS)
Provision the Update Server
with a different version of
software from the current
version running in the UUT.
To perform the phone
program update select the
following phone key
sequence:
SETUP / INSTL / (right arrow)
/ UPDATE / PRGRM / START
Comments
Example:
Server name = phonemgrw2k8.greyhawk.com
Or 192.168.72.60
Server name with file location
appended = phonemgrw2k8.greyhawk.com/teodata/stever
or 192.168.72.60/teodata/stever
(wwwroot is not needed in the
domain name)
Sets the phone update protocol
to HTTP or HTTPS
P/F
You will be using the HTTP file
location in the server for the
update.
The phone will update its
program using HTTPS protocol.
This can be verified as
happening with Wireshark.
The update will take 2 to 4
minutes, depending on the
phone model. Update progress
will be displayed on the phone
display.
Check the error log for
There should be no unexpected SETUP / INSTL / (right arrow) /
problems.
errors in the log.
(right arrow) / LOG / ERROR
Check the phone S/W version It should be the same
SETUP / ADMIN / VERS / S/W.
to see that it matches the
Or select the INFO key on the
version just loaded
phone.
8
8.3.4
INSTALL: UPDATE, PRGRM, TEO
Test on 7810, 7810 + 8030X, 4104, 4101
Step
Action
Expected Result
Comments
Change the Update server
1
address to be the same as the
UCM server.
Configure the phone for TEO
The phone will require a restart. SETUP / INSTL / UPDATE /
2
update protocol.
Select YES when prompted.
PROGRAM / PROTO / TEO
Perform a Update Program
SETUP / INSTL / (right arrow) /
3
UPDATE / PRGRM / START
4
Check the error log for
There should be no unexpected SETUP / INSTL / (right arrow) /
problems after the update is
errors in the log.
(right arrow) / LOG / ERROR
PreReleasePhoneFeatureTest.doc
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P/F
pg. 6
Step
Action
Expected Result
Comments
P/F
complete.
5
8.4 USER: VOICE, VOLUME
Test on 7810, 7810 + 8030X, 4104, 4101
8.4.1
USER: VOICE, VOLUME, Receive
The receive volume setting is for the handset and headset only,
speakerphone receive volume is set during a call with the Volume keys, and also applies to future calls.
Select RCV- or RCV+ to adjust the receiver volume level.
The new setting will be shown in the display.
Step
Action
Expected Result
Comments
P/F
Check the limits of the volume The menu should work as
SETUP / USER / VOICE /
for functionality.
expected.
VOLUME / (HAND, or HEAD)
The volume settings should not
1
distort the audio, and the
volume should be reasonably
usable.
2
8.4.2 USER: VOICE, VOLUME, Transmit
Step
Action
Expected Result
Comments
Check the limits of the
The menu should work as
SETUP / USER / VOICE /
Transmit volume for
expected.
VOLUME / (HAND, HEAD, or
functionality.
The volume settings should not SPKR)
1
distort the audio, and the
volume should be reasonably
usable.
2
8.4.3 USER: VOICE, VOLUME, Reset to Defaults
Step
Action
Expected Result
Set the volume levels to
Levels should all be reset to
1
values other than default, and level 4.
select the Reset key
2
Comments
SETUP / USER / VOICE /
VOLUME / RESET
P/F
P/F
8.5 INSTALL: QOS, Quality of Service
QoS uses diffserv values
Differentiated Services or DiffServ is a computer networking architecture that specifies a simple,
scalable and coarse-grained mechanism for classifying and managing network traffic and providing
Quality of Service (QoS) on modern IP networks. DiffServ can, for example, be used to provide lowlatency to critical network traffic such as voice or streaming media while providing simple best-effort
service to non-critical services such as web traffic or file transfers.
DiffServ uses the 6-bit Differentiated Services Code Point (DSCP) field in the IP header for packet
classification purposes. Explicit Congestion Notification occupies the least-significant 2 bits.
In theory, a network could have up to 64 (i.e. 26) different traffic classes using different markings in the
DSCP. The DiffServ RFCs recommend, but do not require, certain encodings. This gives a network
operator great flexibility in defining traffic classes. In practice, however, most networks use the following
commonly-defined Per-Hop Behaviors:
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pg. 7
Default PHB (Per hop behavior)—which is typically best-effort traffic
Expedited Forwarding (EF) PHB—dedicated to low-loss, low-latency traffic
Assured Forwarding (AF) PHB—gives assurance of delivery under prescribed conditions
Class Selector PHBs—which maintain backward compatibility with the IP Precedence field.
For more info, refer to Wikipedia, where the above info came from.
There are 2 sets of DSCP values. They are not related except for using diffserv values.
1. The default values for QoS signal and voice (i.e. 46, 0x2e hex).
2. The values for MLPP priority calling used only with a supporting server.
The phone looks at the resource priority header for indication of MLPP call.
MLPP is tested in a later test.
The QoS Voice and Signal default settings are 46 (0x2e hex) for non-MLPP calls. This shows up in a
Wireshark Trace for both Signaling and Voice packets when using a non-MLPP server. On an MLPP call,
the signal setting remains wherever it is set, and the voice setting changes to reflect the priority of the
MLPP call.
For MLPP calls, refer to the section on MLPP in this test procedure.
Conversions for priority levels, if needed during test:
Priority Level
Priority Prefix
Hex number
0
28
Override
90
29
Flash
91
2b
Immediate
92
2d
2e
Priority
93
2f
Routine
94
31
3f
Decimal number
0
40
41
43
45
46
47
49
63
Comment
Default QoS
QoS for Signal: look in the SIP Invite in the Internet Protocol, Differentiated Services Field, DSCP value.
The value in parenthesis is the setting in the phone for SIGNAL. It (hex) should match the QoS Signal
setting (dec), located in the menu at SETUP / INSTL / QoS / L3 / SIGNAL.
i.e. Differentiated Services Field: 0xb8 (DSCP 0x2e: Expedited Forwarding; ECN: 0x00)
0x28 hex = 40 decimal
QoS for Voice: look in a RTP packet in the Internet Protocol, Differentiated Services Field, DSCP value.
The value in parenthesis is the setting in the phone for VOICE.
It (hex) should match the QoS Voice setting (dec), located in the menu at SETUP / INSTL / QoS / L3 /
VOICE. For a normal call, the DSCP value will be 49. For a MLPP call, the value will match the priority
level (41, 43, 45, 47, 49).
QoS Voice Setting
46
41
41
43
45
QoS Signal Setting
40 (28 hex)
40 (28 hex)
40 (28 hex)
40 (28 hex)
40 (28 hex)
47
49
40 (28 hex)
40 (28 hex)
PreReleasePhoneFeatureTest.doc
Priority Prefix
None
90, Override
90, Override
91, Flash
92, Immediate
93, Priority
94, Routine
8/28/2014
Invite DSCP
2e
29
29
2b
2d
2f
31
RTP DSCP
2e
29 (Override)
29 (Override)
2b (Flash)
2d (Immediate)
2f (Priority)
31 (Routine)
pg. 8
QoS Voice Setting
QoS Signal Setting
Priority Prefix
Invite DSCP
RTP DSCP
Test on 7810, 7810 + 8030X, 4104
Range=0-63 (decimal) default: voice=49, signal=40
The entire DSCP number appears as 0x2e, which is a 6 bit number; third and 4 th values
are the DSCP value (hex).
Step
1
2
3
4
5
6
Action
Verify that the QoS settings
are set at default.
Expected Result
Default:
Voice=49
Signal=40
Run Wireshark
Call between two phones,
then hang up the call
Stop Wireshark
Comments
SETUP / INSTL / QoS / L3 /
VOICE or SIGNAL
P/F
QoS for Signal: look in the SIP
Invite in the Internet Protocol,
Differentiated Services Field,
DSCP value.
QoS for Voice: look in a RTP
packet in the Internet Protocol,
Differentiated Services Field,
DSCP value.
Voice=49
Signal=40
Change the values to 0 and
retest
Change the values to 63 and
retest
The DSCP value will be 3f(hex).
7
8.6 CODEC TESTING
8.6.1
INSTALL: Keys, Edit Line Key, CODEC
Server in bypass mode. This test tests Codec functionality.
The CODEC menu is used to set the phone’s codec priorities, used in negotiations with the server, or
between phones to select the codec that will be used for each call, as it occurs. There are 4 codecs, and
1 packet RATE that can be set in the phone.
G.711 u-law: Used in the United States and Japan, Pulse Code Modulation, 14 bit resolution.
G.711 a-law: Used in the rest of the world, Pulse Code Modulation, 13 bit resolution.
G.729: lower speech bandwidth. Conserves bandwidth and sacrifices audio quality.
G.722: wider speech bandwidth. Superior audio quality and clarity.
Codecs are numbered 1-4 in the codec menu, and one PACKET RATE is selectable and applies to all
codecs used. The server (if not in bypass mode) tells the phone which CODEC it will be using for each
call, based on codecs supported in the server and the other phone in the call. If the server is in bypass
mode, the phones negotiate the CODEC, and packet RATE themselves.
The phone may not use the codecs that have been selected in the calling phone. If one phone is set at
711, and the other is set at 729, and you place a call between them, the call will not have audio if in
bypass mode.
This test is done in server bypass mode, with the phones determining Codec and packet Rate.
There are several ways to verify test results:
Syslog (in QOS mode) to verify codec settings.
PreReleasePhoneFeatureTest.doc
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pg. 9
Admin/Diag/Packet/Active, during a call to confirm codec settings through the menu.
Wireshark trace.
Test on 7810, 4104, 4101.
Step
Action
Expected Result
Comments
On the UUT select SETUP /
The Line ID will blink red/green Select CODECS G711, G729,
INSTALL / KEYS / (select the and the 1=G711 u-LAW menu and G722, one at a time. Set
primary line key) / EDIT /
will appear, with the selection
CODECS 2 and 3 to NONE for
CODEC, and set CODEC 1 to for CODEC.
each test. Also set packet
each of the CODEC settings
RATES 10, 20, 30, and 40ms
in the calling phone, and
for each CODEC.
1
packet RATES, one at a time
Use these test settings for
and perform the following test
CODEC 1 and corresponding
steps.
packet RATE:
G.711 u-LAW, G.711 a-LAW,
G.729, G722 with Packet
settings 10-40ms at each
setting.
Make a call between the UUT Both phones will use the same The CODECS will appear in the
and another phone while
CODEC (the one selected as
correct order in the call INVITE.
monitoring a Wireshark trace. CODEC 1 on the calling phone), RTP packets from both phones
and the phones will have a
will be using the same CODEC.
voice path.
The CODECS will appear in the Test all of the settings as
call INVITE to the server.
described in the setup in the
The time between packets, for first step.
2
the same phone in the
Wireshark trace, will match the
packet RATE setting made in
the calling phone for the
CODEC selected.
The packet rate as shown in the
Wireshark trace will be
consistent between packets.
3
Re-configure both phones
Default Codec settings are:
used, back to the default
1=G711 u-LAW
settings.
2=G729
3=G722
Packet rate=20ms
P/F
4
8.6.2
AUTO CODEC SELECTION: Install, Keys, Edit Line Key
Server in Bypass mode. This tests automatic Codec selection.
The calling phone determines the priority of the Codecs used in the call. The first Codec in the calling
phone that is supported by the called phone will be used.
This test tests combinations of Codecs and packet Rates to verify the control of the calling phone in
server bypass mode. Bypass mode is when the phone initiating the call, and not the server, is
controlling the Codec and packet Rate used in the call, and the phones are talking to each other directly.
When in bypass mode, if one phone is set at 711, and the other is set at 729, and you place a call
between them, the call will not have audio. In non-bypass mode, the phone will negotiate with the server
and the server will select a codec that results in an audio path being established between the phones,
regardless of the setting. If the phones don't have mutual codecs, the server will translate between
codecs for each phone so that the call can be completed.
PreReleasePhoneFeatureTest.doc
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pg. 10
An easy way to do this test is to use Syslog (QOS setting) to verify codec settings, or use
Admin/Diag/Packet/Active, during a call to confirm codec settings.
Step
1
2
3
4
5
6
7
Test on 7810, 4104, 4101
Action
Set the CODEC/packet RATE
in the phones as follows for
phones A and B.
A CODEC 1: G.711 u-law
A CODEC 2: NONE
A CODEC 3: NONE
B CODEC 1: G.711 u-law
B CODEC 2: NONE
B CODEC 3: NONE
PACKET RATE: 20ms
A calls B
Take a Wireshark trace of the
call.
Repeat the test for packet
rates of 10-40ms.
Expected Result
Codec and packet Rate will be
set as defined in the setup.
The Wireshark trace will show
that Codec will be G.711 u-law,
and the Packets will be 20ms
apart.
Comments
Phone A is the calling phone.
Phone B is the called phone.
To set the Codec and packet
Rate select: SETUP / INSTL /
KEYS / (select a primary line
key) / EDIT / CODEC
P/F
Packet rates displayed on the
Wireshark trace will reflect the
packet settings.
Set the CODEC/packet RATE Codec and packet Rate will be
in the phones as follows for
set as defined in the setup.
phones A and B.
The Wireshark trace will show
A CODEC 1: G.711 a-law
that Codec will be G.711 a-law,
A CODEC 2: NONE
and the Packets will be 20ms
A CODEC 3: NONE
apart.
B CODEC 1: G.711 a-law
B CODEC 2: NONE
B CODEC 3: NONE
PACKET RATE: 20ms
A calls B
Take a Wireshark trace of the
call.
Repeat the test for packet
Packet rates displayed on the
rates of 10-40ms.
Wireshark trace will reflect the
packet settings.
Set the CODEC/packet RATE Codec and packet Rate will be
in the phones as follows for
set as defined in the setup.
phones A and B.
The Wireshark trace will show
A CODEC 1: G.729
that Codec will be G.729, and
A CODEC 2: NONE
the Packets will be 20ms apart.
A CODEC 3: NONE
B CODEC 1: G.729
B CODEC 2: NONE
B CODEC 3: NONE
PACKET RATE: 20ms
A calls B
Take a Wireshark trace of the
call.
Repeat the test for packet
Packet rates displayed on the
rates of 10-40ms.
Wireshark trace will reflect the
packet settings.
Set the CODEC/packet RATE Codec and packet Rate will be
in the phones as follows for
set as defined in the setup.
PreReleasePhoneFeatureTest.doc
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pg. 11
Step
8
Action
phones A and B.
A CODEC 1: G.722
A CODEC 2: NONE
A CODEC 3: NONE
B CODEC 1: G.722
B CODEC 2: NONE
B CODEC 3: NONE
PACKET RATE: 20ms
A calls B
Take a Wireshark trace of the
call.
Repeat the test for packet
rates of 10-40ms.
9
Re-configure both phones
used, back to the default
settings.
Expected Result
The Wireshark trace will show
that Codec will be G.722, and
the Packets will be 20ms apart.
Comments
P/F
Packet rates displayed on the
Wireshark trace will reflect the
packet settings.
Phone default settings:
Codec 1=G.711 u-law
Codec 2=G.729
Codec 3=G722
Packet Rate=20ms
10
8.7 JITTER Test
8.7.1 JITTER, FIXED: INSTALL, KEYS, Edit Line Key
Step
Action
Expected Result
Select an existing line key,
The Line ID will blink red/green
and select EDIT / JTR
and the JITTER BUFFER
1
ADAPTIVE will appear, with
selections for DELAY and
FIXED.
Select FIXED.
The Line ID will blink red/green
and the JITTER
2
FIXED,DELAY=35 menu will
appear, with the selections:
DLY+, DLY-, and ADAPT
Adjust the jitter delay through The settings will all be
3
the settings to verify that they accepted.
are all accepted.
Re-configure the phone back
4
to its default settings
5
Comments
The default delay for FIXED
should be JITTER
FIXED,DELAY=35.
Fixed Jitter Range= 10-90
Default=35
Default = ADAPT
8.7.2 JITTER, ADAPTIVE: INSTALL KEYS, Edit Line Key
Step
Action
Expected Result
Comments
Select an existing line key,
The Line ID will blink red/green
and select EDIT / JTR /
and the JITTER BUFFER
1
ADAPT
ADAPTIVE will appear, with
selections for DELAY and
FIXED.
Select DELAY
The DELAY: MIN=10 MAX=100 The default Adaptive Jitter
display will appear with the
Delay range is 10-100ms.
2
selections: MIN+, MIN-, MAX+, Default=10
MAX-
PreReleasePhoneFeatureTest.doc
8/28/2014
P/F
P/F
pg. 12
Step
3
4
Action
Adjust the MIN and MAX
DELAY through their ranges
to verify that they are all
accepted.
Re-configure the phone back
to its default settings
Expected Result
Settings will all be accepted.
Comments
MIN range=0-280
MAX range=0-300
P/F
The default Adaptive Jitter
Delay range is 10-100ms.
Default=10
5
8.8 USER: RING, Personal Ringing
Test on 7810, 4104, 4101
Personal Settings: Tone, Offhook, Control, Local Call Forward (LCFWD)
LCFWD sets the timeout for Call Forward NO ANS
8.8.1 USER: RING, Personal Ringing, TONE
Step
Action
Expected Result
Check the default ring tone
Default tone=5
1
2
3
Try all of the ring tone
selections.
Call the phone at each ring
tone selection to verify that
the ring tones work when the
phone is called.
Comments
Setup / User / Ring / Tone / 1-6
After a phone Reset, the ring
tone default is 5.
There will be ring tones for keys When loaded with the xml file,
1-6, but not for 7-9
ring tone 8 loads and displays
as 8, but the phone rings at
tone 6.
The tones will reflect the
settings.
P/F
4
8.8.2
OFFHOOK Ringing
This controls the phone ringing when offhook. SINGLE or NORMAL, default=SINGLE
Step
Action
Expected Result
Comments
Check that the default setting
SETUP / USER / RING /
1
is SINGLE
OFFHOOK / SINGLE
Take the handset offhook,
The offhook phone should only
2
and then place a call to the
ring once, and should be at a
offhook phone.
reduced volume from normal.
3
Hang up the phones.
Change the setting to
SETUP / USER / RING /
4
NORMAL.
OFFHOOK / NORMAL
Take the handset offhook,
The offhook phone should ring
5
and then place a call to the
continuously, and be at a
offhook phone.
reduced volume from normal.
6
8.8.3
P/F
RING CONTROL
Individual line appearances can be set to always ring, never
ring, or ring after a delay. SETUP / USER / RING / CONTROL / (select a line key) / ALWAYS, NEVER,
WAIT2-7
The 4101 has no ring control; only test 7810 and 4104.
PreReleasePhoneFeatureTest.doc
8/28/2014
pg. 13
8.8.4 Always Ring
Step
Action
Set to Always Ring
Expected Result
1
Call the phone.
2
The phone should ring
continuously, and the line key
should blink green.
Comments
SETUP / USER / RING /
CONTROL / (select a line key) /
ALWAYS
Always Ring is the default.
On Teo UCM the call will get
transferred to voicemail after 4
rings.
P/F
3
8.8.5
Step
1
2
3
Never Ring
The 4101 has no ring control; only test 7810 and 4104.
Action
Expected Result
Comments
Set line 1 to NEVER ring and The phone will not ring, but the With a line set to NEVER ring
call the phone.
line key will light.
and USER / PREF set to
RINGING, with an incoming
call, when you pick up the
handset, the phone will select
an IDLE line, not the ringing
line, because the phone needs
to have ringing to pick up a
ringing line.
Make a call from line 1 to
another phone, and answer
the call.
Call the phone (line 2)
Line 2 will light, but no ring will On Teo UCM the call will get
be heard.
transferred to voicemail after 4
rings.
Ring after a Delay
The 4101 has no ring control; only test 7810 and 4104.
Step
Action
Expected Result
Comments
Check Delay 2 and Delay 7
On delay 2, the phone will not On Teo UCM the call will get
ring for the first 2 rings, and
transferred to voicemail after 4
then will begin ringing.
rings.
1
On delay 7, the call will be
transferred to voicemail before
the phone gets to the 7thring.
2
P/F
8.8.6
P/F
8.9 Adjusting the Ringer Volume
Test on 7810, 7810 + 8030X, 4104, 4101
The Volume keys adjust the ringer volume while you are not on a call.
The telephone will ring once with the new ringer volume setting, and
the new setting will be shown briefly in the display.
When the volume is at the lowest setting, the phone will display
"RINGER OFF" when idle.
Press the Volume Up key while you are not on a call to enable
the ringer.
PreReleasePhoneFeatureTest.doc
8/28/2014
pg. 14
Step
1
2
3
4
Action
Select the ring volume up
arrow repeatedly and verify
that the volume increases
with each press, until it
reaches the maximum
volume level.
Expected Result
Comments
Ring volume will increase each The beep lasts for about 2
time the volume up arrow is
seconds each time the volume
pressed, until the phone
up/down arrow is pressed.
reaches maximum volume.
Then the phone will still beep
when the up arrow is pressed,
but the volume will not increase
any more.
P/F
Call the phone and verify that
the phone rings with
maximum volume.
Select the ring volume down
arrow repeatedly and verify
that the volume decreases
with each press, until it
reaches the minimum volume
level (OFF).
Ring volume will decrease each
time the volume down arrow is
pressed, until the phone
reaches minimum volume (off).
Then the phone will continue to
not beep when the down arrow
is pressed.
Call the phone and verify that The phone will not ring, but the
the phone does not ring.
line key LED will still flash
green.
5
8.9.1
Distinctive Ringing of SIP Phone
Test on 7810, 4104, 4101
Tested on TEO-UC141 with an incoming pstn call, and an inside call.
Tested on Lucent for pre-emption
Step
Action
Expected Result
1
B goes offhook
B receives dialtone
A rings with inside ring tone, B
2
B dials A
has ringback
3
A goes offhook
A and B confirm talk path
4
B goes onhook
A in idle state
5
pstn dials A
A rings with outside ring tone
6
A goes offhook
A and pstn confirm talk path
7
C goes onhook
A in idle state
8
Comments
P/F
8.10 USER: DIR, User Directory
The User directory is available through the User/Dir menu selection, or by selecting the DIR key on the
keyboard.
Test on 7810, 7810 + 8030X, 4104, 4101
USER: DIR, Immediate or Editable Dialing
For one-touch dialing, enter a # character at the end of the dial string.
The string will be dialed immediately when the DIR key is selected. Without the #, when the directory
entry is selected, a line becomes active, and the number is dialed after the dial timeout expires. This
allows you to enter more digits after selecting the DIR entry.
Test on 7810, 7810 + 8030X, 4104, 4101.
Step
Action
Expected Result
Comments
P/F
1
Create call directory entries
To make a new DIR entry
with and without the # at the
select:
PreReleasePhoneFeatureTest.doc
8/28/2014
pg. 15
Step
Action
end of the number
Expected Result
Place calls with a number that
has the # at the end of the
number.
2
3
4
5
6
7
8
9
10
Comments
DIR / EDIT / (select an unused
key) / NUMBER / (enter a
number to dial) / OK / NAME /
(enter a name for the DIR entry)
/ OK. Then exit the DIR. Dial
Timeout is set at location
SETUP / INSTL / CALL / DIAL.
Default= 10 seconds.
There are 2 ways to dial from
the Directory.
1. pick up the handset first,
press the DIR key and then
select a dir entry.
2. Press the DIR key, select an
entry, and then pick up the
handset.
P/F
Verify the talk path and then
end the call.
Place calls with a number that You will have to wait for the dial
does not have the # at the
timeout to end, or press the
end of the number.
SEND softkey to complete
dialing the number.
Verify the talk path and then
end the call.
Create a Directory entry with The Dir entry will appear to be This is to enable the user to
a NUMBER and no NAME
blank, but will dial the number create hidden directory entries.
when selected.
Hidden directory entries always
sort to the end of the visible
directory.
Dial the hidden directory entry The number will dial.
Verify the talk path.
End the call.
8.11 Call Log
Stores up to 100 calls, first in, last out.
Us Asterisk for the load test, test, or an engineering server, do not use the house phone system. The test
runs from TCS Logview.
Run a load test to place calls into the call log of three phone types, 7810, 4104, and 4101.
A Phone RESTART clears the call log; the Error log must be cleared manually through the menus.
TO DUPLICATE:
Bring up 2 phones with no logs enabled. Connect to both phones through the Telnet; serial port can be
used. Place a call to a different phone than the ones used in the test, as a marker in the log, since all of
the other logs will be for the phone numbers used during the test.
At telnet port on the callee phone, enter the following:
CeTest
at1
ce1,100
At the telnet port on the caller phone, enter the following:
PreReleasePhoneFeatureTest.doc
8/28/2014
pg. 16
CeTest
at1
cr1049#,100,100
Note "1049#" is the number of the target phone. The test will run for 100 calls.
If you need to end the test for some reason, press the Mute key on the phone. Disconnect Logview, clear
the Error Log, and Restart the phone to recover memory and clear the Call Log.
Test on 7810, 4104, 4101.
Step
Action
Expected Result
Comments
P/F
Restart a 4101, 4104, and a
The call logs will be cleared on Check the call log before and
1
7810
each phone
after the Restart to verify that
the log changed
Start a load test using the
Start the callee first. Then start
2
7810 as the callee, and the
the callers.
other two phones as callers.
make a call from each of the
callers to a phone not used in
3
the test, to be used as a
placeholder in the log.
Do the same for the callee.
Let the test run for at least
4
100 caller calls on each
caller.
Count the number of calls
The logs should display a
logged in each phone log.
message on both ends of the
5
log, stating NO NEWER
OUTGOING CALLS, OR NO
OLDER OUTGOING CALLS
6
8.12 HOT/WARM Dialing
Line Mode HOT/WARM/NORMAL
This is configured in the xml file as general settings.
Default=NORMAL
<line_mode>WARM</line_mode>
<hot_warm_username>1065</hot_warm_username>
<warmline_timeout>5</warmline_timeout>
Step
Action
Expected Result
A receives dialtone
1
A keys *96+4 digit #
C alerts to intercom
A and C confirm talk path
C goes onhook
A in idle state
2
3
4
Comments
A must have Intercom COS bit
set
xxxx is extension of phone C
that supports intercom, or has
Voice Announce enabled.
P/F
Comments
SETUP / ADMIN / INSPCT
P/F
8.13 ADMIN: INSPCT, Inspect Keys
Step
1
Test on 7810, 7810 + 8030X, 4104, 4101
Action
Expected Result
Inspect all key types.
Types should all be reported
correctly
2
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pg. 17
8.14 INSTALL: PASSWD, Installation Password
Password was change to PIN with version 0x.03.15.15.
<install_pin></install_pin>
The PIN is defined as 4 to 20 numbers, no alphabet characters or special characters.
After the PIN is entered one time, it has to be entered every time after that.
The PIN can only be deleted by Resetting the phone, or entering a null pin with the xml tag.
Test on 7810, 7810 + 8030X, 4104, 4101
Step
Action
Expected Result
Comments
Using the menu, set a PIN of
1
4 numbers and verify that it
works.
Using the menu, set a PIN of
2
20 numbers and verify that it
works.
Test the boundaries.
5 and 21 numbers, Alpha
3
characters, special characters,
null.
Verify that once you have put
The only way to clear the PIN is
4
in a PIN, that you have to use
to Reset the phone, or enter a
a PIN after that.
PIN of null with the xml file.
Use the xml file to null out the The install pin should be
5
install pin.
removed.
6
P/F
8.15 USER: PIN, Setting a Call Log PIN
Test on 7810, 4104, 4101
Original Password Requirements
Passwords must be at least 9 characters long, and must
contain at least :
2 digits
2 upper case letters
2 lower case letters
2 special characters
The password cannot contain the line ID, forward or backward.
At least 4 characters must have changed from previous password entries.
Select HELP to show these password requirements on screen.
Press the OK key.
Repeat the password when prompted to verify the new entry.
Press the OK key.
“PASSWORD SET” will be displayed to confirm the new password.
Record your password for future reference.
Press the OK key to return to the User Options menu or press
the SETUP key to exit Setup Mode.
The password requirements are now limited to a range of 4 to 20 numbers for all phone types, and there
is no Help screen.
Step
Action
Expected Result
Comments
P/F
1
Set PIN for the User Log.
Use the PIN to verify that they
2
were set.
3
Clear the PIN.
PreReleasePhoneFeatureTest.doc
8/28/2014
pg. 18
Step
4
5
6
7
Action
Enter the Log to verify that
the passwords were cleared
Try alpha characters and
special characters.
Check the limits of the PIN.
Expected Result
Comments
P/F
Only numbers will be allowed.
4-20 numbers
8.16 INSTALL: IP, IP Addresses, Syslog
In order to test the Syslog function, use your PC address as the Syslog server address, and run the
TFTPD32 program on the PC, with Syslog enabled.
<ip_syslog source=”DHCP4”>192.168.72.133</ip_syslog>
Logging option for the Syslog Server operation.
<syslog_option>NONE</syslog_option>
Valid options are,
NONE – Disabled / No Logging (Default)
BASIC – SIP Phone Error Logs are sent to Syslog server
QOS – Error Logs and Quality of Service call packet statistics sent.
In the 4101, you can only set the Syslog IP address in the phone menu, and the Option defaults to NONE.
To set the Syslog Option to anything besides NONE, you have to set the Syslog Option in the xml file,
using the tag:
<syslog_option>NONE</syslog_option>
NONE – Disabled / No Logging (Default)
BASIC – SIP Phone Error Logs are sent to Syslog server
QOS – Error Logs and Quality of Service call packet statistics sent.
Step
Action
Expected Result
Comments
P/F
Set up a Syslog server on the
Run the TFTPD32 program on
1
Test PC
the PC, with the Syslog option
enabled.
Set the Syslog Option menu
Verify that no logs are sent to
selection to NONE.
the Syslog server when the
2
phone is restarted, or when
calls are made.
Set the Syslog Option menu
Verify that SIP Phone Error logs
3
selection to BASIC.
re sent to the Syslog server.
Set the Syslog Option menu
Verify that Error Logs and
selection to QOS.
Quality of Service call packet
4
statistics are sent.
5
9 ISSUES:
PreReleasePhoneFeatureTest.doc
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pg. 19
(This test must be run at least once for each release.)
Test Date:
Tester:
Phones Tested:
Platforms:
Phone Test version:
Final Version No:
Particular attention was focused on phone features
This version is for use on Teo UCM, Avaya, Redcom, BroadSoft, Cisco, NEC
TABLE OF CONTENTS
CONTENT
1
2
3
4
5
6
7
8
8.1
8.2
8.3
8.3.1
8.3.2
8.3.3
8.3.4
8.4
8.4.1
8.4.2
8.4.3
8.5
8.6
8.6.1
8.6.2
8.7
8.7.1
8.7.2
8.8
8.8.1
8.8.2
8.8.3
PAGE NUMBER
Scope.................................................................................................................................................... 2
Hardware used for the test.................................................................................................................... 2
Test Servers.......................................................................................................................................... 2
Software versions used during the test................................................................................................. 2
Test Responses..................................................................................................................................... 2
Test Failures.......................................................................................................................................... 3
Order of Test......................................................................................................................................... 3
Regression Tests................................................................................................................................... 3
INSTALL: RESET, MAC xml.................................................................................................................. 3
INSTALL: RESET, LINE xml.................................................................................................................. 4
PROGRAM UPDATE............................................................................................................................ 4
INSTALL: UPDATE, PRGRM, TFTP Program Update..........................................................................4
INSTALL: UPDATE, PRGRM, Program Update Not Required..............................................................5
INSTALL: UPDATE, PRGRM, HTTP or HTTPS Program Update.........................................................5
INSTALL: UPDATE, PRGRM, TEO....................................................................................................... 6
USER: VOICE, VOLUME...................................................................................................................... 7
USER: VOICE, VOLUME, Receive....................................................................................................... 7
USER: VOICE, VOLUME, Transmit...................................................................................................... 7
USER: VOICE, VOLUME, Reset to Defaults........................................................................................ 7
INSTALL: QOS, Quality of Service........................................................................................................ 7
CODEC TESTING................................................................................................................................. 9
INSTALL: Keys, Edit Line Key, CODEC................................................................................................ 9
AUTO CODEC SELECTION: Install, Keys, Edit Line Key...................................................................11
JITTER Test......................................................................................................................................... 12
JITTER, FIXED: INSTALL, KEYS, Edit Line Key................................................................................12
JITTER, ADAPTIVE: INSTALL KEYS, Edit Line Key...........................................................................13
USER: RING, Personal Ringing.......................................................................................................... 13
USER: RING, Personal Ringing, TONE.............................................................................................. 13
OFFHOOK Ringing............................................................................................................................. 13
RING CONTROL................................................................................................................................ 14
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pg. 1
8.8.4
8.8.5
8.8.6
8.9
8.9.1
8.10
8.11
8.12
8.13
8.14
8.15
8.16
9
Always Ring........................................................................................................................................ 14
Never Ring.......................................................................................................................................... 14
Ring after a Delay............................................................................................................................... 14
Adjusting the Ringer Volume............................................................................................................... 15
Distinctive Ringing of SIP Phone........................................................................................................ 15
USER: DIR, User Directory................................................................................................................. 16
Call Log............................................................................................................................................... 17
HOT/WARM Dialing............................................................................................................................ 18
ADMIN: INSPCT, Inspect Keys.......................................................................................................... 18
INSTALL: PASSWD, Installation Password.........................................................................................18
USER: PIN, Setting a Call Log PIN..................................................................................................... 18
INSTALL: IP, IP Addresses, Syslog..................................................................................................... 19
ISSUES:.............................................................................................................................................. 20
1 Scope
An engineering regression test to test common phone functions, including call control.
Testing is Ad-Hoc unless specific test steps are listed. In the case of tests with well developed steps,
there is still Ad-Hoc testing performed.
2 Hardware used for the test
7810
7810-TSG
7810 + 8030X
4104
4101
3 Test Servers
Teo UCM
4 Software versions used during the test
Phone Software: v0x.04.20
5 Test Responses
√, P: Pass
Х, F: Fail
N/T: Not Tested
N/A: Not Applicable
BUG: Issues that occurred during the test.
PreReleasePhoneFeatureTest.doc
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pg. 2
6 Test Failures
Test Failures and reported issues are documented as Bugs in the JIRA Database, along with bug repair
and retest results. Test failures are repaired in following software versions that may or may not be part of
this final release. Bug repair priorities are determined by engineering management. To review open bugs
with lower priorities, review the JIRA Database. A basic understanding of Teo VOIP phone operation is
necessary in order to complete this test, and a focus on ad-hoc phone testing (unusual actions with a
specific focus) is also desirable to find additional problems. Tests are added as new scenarios are
discovered.
Test as appropriate on 7810, 7810 + 8030X, 7810-TSG, 4104, 4101.
7 Order of Test
This list is provided as a logical progression of events and tests to be performed.
Test
Test New Features or Bug Fixes
Program Update from the new version
to something else
Handset and Speaker volume
Voice quality test
QoS Test
Codec and P time test
Ring Controls and ringer Volume
Dialing from the Directory and Call
Log
Hot/Warm Dialing
Admin Inspect Keys
Install PIN
User PIN
Syslog Test
Time Estimate
PASS/FAIL
8 Regression Tests
8.1 INSTALL: RESET, MAC xml
Test on 7810, 7810 + 8030X, 4104, 4101
Step
1
2
Action
Configure the TCS7000A.xml
file (global xml tag file) with a
configuration tag set for MAC
address.
Expected Result
Create a unique configuration
xml file for the test phone,
with the file name made up of
PreReleasePhoneFeatureTest.doc
8/28/2014
Comments
Set the TCS7000A config tag
to MAC:
<config_file_name_base>MA
C</config_file_name_base>
Or comment out the line;
MAC is the default
i.e. 00048DFFFFE8.xml
P/F
pg. 3
Step
Action
the phone MAC address.
Reset the phone, and watch
the TFTP server window
3
Expected Result
The phone will load the
TCS7000A.xml file and its
MAC.xml file.
The phone will initialize and
register.
Comments
P/F
As the phone initializes, it will
prompt the user for needed
data.
4
8.2 INSTALL: RESET, LINE xml
Test on 7810, 7810 + 8030X, 4104, 4101
Step
1
2
Action
Configure the TCS7000A.xml
file (global xml tag file) with a
configuration tag set for LINE
address.
Create a unique configuration
xml file for the test phone,
with the file name made up of
the phone Line ID.
Reset the phone, and watch
the TFTP server window.
3
Expected Result
The phone will load the
TCS7000A.xml file and its
Line ID.xml file.
The phone will initialize and
register.
Comments
Set the TCS7000A config tag
to LINE:
<config_file_name_base>LIN
E</config_file_name_base>
P/F
As the phone initializes, it will
prompt the user for needed
data.
4
8.3 PROGRAM UPDATE
8.3.1
Step
1
2
3
4
INSTALL: UPDATE, PRGRM, TFTP Program Update
Test on 7810, 7810 + 8030X, 4104, 4101
Action
Expected Result
Provision the Update Server
with a different version of
software from the current
version running in the UUT.
Configure the UUT for TFTP
update
Comments
P/F
SETUP / INSTL / (right arrow) /
UPDATE / PRGRM / PROTO /
TFTP
SETUP / INSTL / (right arrow) /
UPDATE / PRGRM / START
Select "UPDATE / PRGRM"to The UUT should load the
load the different software
different software, and
from the Update Server.
RESTART. After RESTART the
phone should successfully
register, and have the same
LINE ID and other settings as
before the update.
Check the error log for
There should be no unexpected SETUP / INSTL / (right arrow) /
problems.
errors in the log.
(right arrow) / LOG / ERROR
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8/28/2014
pg. 4
Step
5
Action
Expected Result
Check the phone S/W version It should be the new version.
to see that it matches the
version just loaded
Comments
P/F
SETUP / ADMIN / VERS / S/W
6
8.3.2
INSTALL: UPDATE, PRGRM, Program Update Not Required
Test on 7810, 7810 + 8030X, 4104, 4101
Step
Action
Expected Result
Comments
Provision the Update Server
SETUP / ADMIN / VERS / S/W
with the same version of
1
software that is presently
running in the phone.
Configure the UUT for TFTP
SETUP / INSTL / (right arrow) /
2
update
UPDATE / PRGRM / PROTO /
TFTP
Perform a Update Program to The UUT should not load the
SETUP / INSTL / (right arrow) /
attempt to load the same
software into the phone. The UPDATE / PRGRM / START
3
software from the Update
phone will display:
Server.
PROGRAM UPDATE NOT
REQUIRED
Check the error log for
There should be no unexpected SETUP / INSTL / (right arrow) /
4
problems.
errors in the log.
(right arrow) / LOG / ERROR
5
INSTALL: UPDATE, PRGRM, HTTP or HTTPS Program Update
Test on 7810, 7810 + 8030X, 4104, 4101
Step
Action
Expected Result
Comments
Create a location in the
Server name = phonemgrcomputer Update server path,
w2k8.greyhawk.tonecommande
for program update files. This
r.com
will be a different location
Server name with file location
than for TFTP updates.
appended = phonemgrThis path is where the
w2k8.greyhawk.tonecommande
digitally signed files are
r.com/teotestdata/stever.
located under the update
This is the location that the
server root directory.
HTTPS server is pointed at.
Place the folder containing
1
the phone update program
If you are connected to a TEO
files in the HTTP/HTTPS
UCM server, Program Updates
update file address. Also
are done with HTTPS protocol.
place all of the folder contents
This is adequate for this test.
in the same location.
P/F
8.3.3
2
Or do a Program Update from
the keypad on a phone
connected to a TEO UCM
server.
Edit the phone's UPDATE IP
address (Domain Name),
appending the location for the
PreReleasePhoneFeatureTest.doc
P/F
The program update file
location that was just created.
8/28/2014
pg. 5
Step
3
4
5
6
7
Action
program Update files:
On the phone select SETUP /
INSTL / IP / (right arrow) /
UPDATE / EDIT / DHCP4 /
OK.
Use the navigation arrows to
move the cursor to the end of
the displayed Update Server
Address. Append the
HTTP/HTTPS file location.
Expected Result
Enter the following key
sequence on the phone to set
the phone’s update protocol
to HTTP or HTTPS:
On the phone, select
SETUP / INSTL / (right arrow)
/ UPDATE / PRGRM /
PROTO / HTTP(or HTTPS)
Provision the Update Server
with a different version of
software from the current
version running in the UUT.
To perform the phone
program update select the
following phone key
sequence:
SETUP / INSTL / (right arrow)
/ UPDATE / PRGRM / START
Comments
Example:
Server name = phonemgrw2k8.greyhawk.com
Or 192.168.72.60
Server name with file location
appended = phonemgrw2k8.greyhawk.com/teodata/stever
or 192.168.72.60/teodata/stever
(wwwroot is not needed in the
domain name)
Sets the phone update protocol
to HTTP or HTTPS
P/F
You will be using the HTTP file
location in the server for the
update.
The phone will update its
program using HTTPS protocol.
This can be verified as
happening with Wireshark.
The update will take 2 to 4
minutes, depending on the
phone model. Update progress
will be displayed on the phone
display.
Check the error log for
There should be no unexpected SETUP / INSTL / (right arrow) /
problems.
errors in the log.
(right arrow) / LOG / ERROR
Check the phone S/W version It should be the same
SETUP / ADMIN / VERS / S/W.
to see that it matches the
Or select the INFO key on the
version just loaded
phone.
8
8.3.4
INSTALL: UPDATE, PRGRM, TEO
Test on 7810, 7810 + 8030X, 4104, 4101
Step
Action
Expected Result
Comments
Change the Update server
1
address to be the same as the
UCM server.
Configure the phone for TEO
The phone will require a restart. SETUP / INSTL / UPDATE /
2
update protocol.
Select YES when prompted.
PROGRAM / PROTO / TEO
Perform a Update Program
SETUP / INSTL / (right arrow) /
3
UPDATE / PRGRM / START
4
Check the error log for
There should be no unexpected SETUP / INSTL / (right arrow) /
problems after the update is
errors in the log.
(right arrow) / LOG / ERROR
PreReleasePhoneFeatureTest.doc
8/28/2014
P/F
pg. 6
Step
Action
Expected Result
Comments
P/F
complete.
5
8.4 USER: VOICE, VOLUME
Test on 7810, 7810 + 8030X, 4104, 4101
8.4.1
USER: VOICE, VOLUME, Receive
The receive volume setting is for the handset and headset only,
speakerphone receive volume is set during a call with the Volume keys, and also applies to future calls.
Select RCV- or RCV+ to adjust the receiver volume level.
The new setting will be shown in the display.
Step
Action
Expected Result
Comments
P/F
Check the limits of the volume The menu should work as
SETUP / USER / VOICE /
for functionality.
expected.
VOLUME / (HAND, or HEAD)
The volume settings should not
1
distort the audio, and the
volume should be reasonably
usable.
2
8.4.2 USER: VOICE, VOLUME, Transmit
Step
Action
Expected Result
Comments
Check the limits of the
The menu should work as
SETUP / USER / VOICE /
Transmit volume for
expected.
VOLUME / (HAND, HEAD, or
functionality.
The volume settings should not SPKR)
1
distort the audio, and the
volume should be reasonably
usable.
2
8.4.3 USER: VOICE, VOLUME, Reset to Defaults
Step
Action
Expected Result
Set the volume levels to
Levels should all be reset to
1
values other than default, and level 4.
select the Reset key
2
Comments
SETUP / USER / VOICE /
VOLUME / RESET
P/F
P/F
8.5 INSTALL: QOS, Quality of Service
QoS uses diffserv values
Differentiated Services or DiffServ is a computer networking architecture that specifies a simple,
scalable and coarse-grained mechanism for classifying and managing network traffic and providing
Quality of Service (QoS) on modern IP networks. DiffServ can, for example, be used to provide lowlatency to critical network traffic such as voice or streaming media while providing simple best-effort
service to non-critical services such as web traffic or file transfers.
DiffServ uses the 6-bit Differentiated Services Code Point (DSCP) field in the IP header for packet
classification purposes. Explicit Congestion Notification occupies the least-significant 2 bits.
In theory, a network could have up to 64 (i.e. 26) different traffic classes using different markings in the
DSCP. The DiffServ RFCs recommend, but do not require, certain encodings. This gives a network
operator great flexibility in defining traffic classes. In practice, however, most networks use the following
commonly-defined Per-Hop Behaviors:
PreReleasePhoneFeatureTest.doc
8/28/2014
pg. 7
Default PHB (Per hop behavior)—which is typically best-effort traffic
Expedited Forwarding (EF) PHB—dedicated to low-loss, low-latency traffic
Assured Forwarding (AF) PHB—gives assurance of delivery under prescribed conditions
Class Selector PHBs—which maintain backward compatibility with the IP Precedence field.
For more info, refer to Wikipedia, where the above info came from.
There are 2 sets of DSCP values. They are not related except for using diffserv values.
1. The default values for QoS signal and voice (i.e. 46, 0x2e hex).
2. The values for MLPP priority calling used only with a supporting server.
The phone looks at the resource priority header for indication of MLPP call.
MLPP is tested in a later test.
The QoS Voice and Signal default settings are 46 (0x2e hex) for non-MLPP calls. This shows up in a
Wireshark Trace for both Signaling and Voice packets when using a non-MLPP server. On an MLPP call,
the signal setting remains wherever it is set, and the voice setting changes to reflect the priority of the
MLPP call.
For MLPP calls, refer to the section on MLPP in this test procedure.
Conversions for priority levels, if needed during test:
Priority Level
Priority Prefix
Hex number
0
28
Override
90
29
Flash
91
2b
Immediate
92
2d
2e
Priority
93
2f
Routine
94
31
3f
Decimal number
0
40
41
43
45
46
47
49
63
Comment
Default QoS
QoS for Signal: look in the SIP Invite in the Internet Protocol, Differentiated Services Field, DSCP value.
The value in parenthesis is the setting in the phone for SIGNAL. It (hex) should match the QoS Signal
setting (dec), located in the menu at SETUP / INSTL / QoS / L3 / SIGNAL.
i.e. Differentiated Services Field: 0xb8 (DSCP 0x2e: Expedited Forwarding; ECN: 0x00)
0x28 hex = 40 decimal
QoS for Voice: look in a RTP packet in the Internet Protocol, Differentiated Services Field, DSCP value.
The value in parenthesis is the setting in the phone for VOICE.
It (hex) should match the QoS Voice setting (dec), located in the menu at SETUP / INSTL / QoS / L3 /
VOICE. For a normal call, the DSCP value will be 49. For a MLPP call, the value will match the priority
level (41, 43, 45, 47, 49).
QoS Voice Setting
46
41
41
43
45
QoS Signal Setting
40 (28 hex)
40 (28 hex)
40 (28 hex)
40 (28 hex)
40 (28 hex)
47
49
40 (28 hex)
40 (28 hex)
PreReleasePhoneFeatureTest.doc
Priority Prefix
None
90, Override
90, Override
91, Flash
92, Immediate
93, Priority
94, Routine
8/28/2014
Invite DSCP
2e
29
29
2b
2d
2f
31
RTP DSCP
2e
29 (Override)
29 (Override)
2b (Flash)
2d (Immediate)
2f (Priority)
31 (Routine)
pg. 8
QoS Voice Setting
QoS Signal Setting
Priority Prefix
Invite DSCP
RTP DSCP
Test on 7810, 7810 + 8030X, 4104
Range=0-63 (decimal) default: voice=49, signal=40
The entire DSCP number appears as 0x2e, which is a 6 bit number; third and 4 th values
are the DSCP value (hex).
Step
1
2
3
4
5
6
Action
Verify that the QoS settings
are set at default.
Expected Result
Default:
Voice=49
Signal=40
Run Wireshark
Call between two phones,
then hang up the call
Stop Wireshark
Comments
SETUP / INSTL / QoS / L3 /
VOICE or SIGNAL
P/F
QoS for Signal: look in the SIP
Invite in the Internet Protocol,
Differentiated Services Field,
DSCP value.
QoS for Voice: look in a RTP
packet in the Internet Protocol,
Differentiated Services Field,
DSCP value.
Voice=49
Signal=40
Change the values to 0 and
retest
Change the values to 63 and
retest
The DSCP value will be 3f(hex).
7
8.6 CODEC TESTING
8.6.1
INSTALL: Keys, Edit Line Key, CODEC
Server in bypass mode. This test tests Codec functionality.
The CODEC menu is used to set the phone’s codec priorities, used in negotiations with the server, or
between phones to select the codec that will be used for each call, as it occurs. There are 4 codecs, and
1 packet RATE that can be set in the phone.
G.711 u-law: Used in the United States and Japan, Pulse Code Modulation, 14 bit resolution.
G.711 a-law: Used in the rest of the world, Pulse Code Modulation, 13 bit resolution.
G.729: lower speech bandwidth. Conserves bandwidth and sacrifices audio quality.
G.722: wider speech bandwidth. Superior audio quality and clarity.
Codecs are numbered 1-4 in the codec menu, and one PACKET RATE is selectable and applies to all
codecs used. The server (if not in bypass mode) tells the phone which CODEC it will be using for each
call, based on codecs supported in the server and the other phone in the call. If the server is in bypass
mode, the phones negotiate the CODEC, and packet RATE themselves.
The phone may not use the codecs that have been selected in the calling phone. If one phone is set at
711, and the other is set at 729, and you place a call between them, the call will not have audio if in
bypass mode.
This test is done in server bypass mode, with the phones determining Codec and packet Rate.
There are several ways to verify test results:
Syslog (in QOS mode) to verify codec settings.
PreReleasePhoneFeatureTest.doc
8/28/2014
pg. 9
Admin/Diag/Packet/Active, during a call to confirm codec settings through the menu.
Wireshark trace.
Test on 7810, 4104, 4101.
Step
Action
Expected Result
Comments
On the UUT select SETUP /
The Line ID will blink red/green Select CODECS G711, G729,
INSTALL / KEYS / (select the and the 1=G711 u-LAW menu and G722, one at a time. Set
primary line key) / EDIT /
will appear, with the selection
CODECS 2 and 3 to NONE for
CODEC, and set CODEC 1 to for CODEC.
each test. Also set packet
each of the CODEC settings
RATES 10, 20, 30, and 40ms
in the calling phone, and
for each CODEC.
1
packet RATES, one at a time
Use these test settings for
and perform the following test
CODEC 1 and corresponding
steps.
packet RATE:
G.711 u-LAW, G.711 a-LAW,
G.729, G722 with Packet
settings 10-40ms at each
setting.
Make a call between the UUT Both phones will use the same The CODECS will appear in the
and another phone while
CODEC (the one selected as
correct order in the call INVITE.
monitoring a Wireshark trace. CODEC 1 on the calling phone), RTP packets from both phones
and the phones will have a
will be using the same CODEC.
voice path.
The CODECS will appear in the Test all of the settings as
call INVITE to the server.
described in the setup in the
The time between packets, for first step.
2
the same phone in the
Wireshark trace, will match the
packet RATE setting made in
the calling phone for the
CODEC selected.
The packet rate as shown in the
Wireshark trace will be
consistent between packets.
3
Re-configure both phones
Default Codec settings are:
used, back to the default
1=G711 u-LAW
settings.
2=G729
3=G722
Packet rate=20ms
P/F
4
8.6.2
AUTO CODEC SELECTION: Install, Keys, Edit Line Key
Server in Bypass mode. This tests automatic Codec selection.
The calling phone determines the priority of the Codecs used in the call. The first Codec in the calling
phone that is supported by the called phone will be used.
This test tests combinations of Codecs and packet Rates to verify the control of the calling phone in
server bypass mode. Bypass mode is when the phone initiating the call, and not the server, is
controlling the Codec and packet Rate used in the call, and the phones are talking to each other directly.
When in bypass mode, if one phone is set at 711, and the other is set at 729, and you place a call
between them, the call will not have audio. In non-bypass mode, the phone will negotiate with the server
and the server will select a codec that results in an audio path being established between the phones,
regardless of the setting. If the phones don't have mutual codecs, the server will translate between
codecs for each phone so that the call can be completed.
PreReleasePhoneFeatureTest.doc
8/28/2014
pg. 10
An easy way to do this test is to use Syslog (QOS setting) to verify codec settings, or use
Admin/Diag/Packet/Active, during a call to confirm codec settings.
Step
1
2
3
4
5
6
7
Test on 7810, 4104, 4101
Action
Set the CODEC/packet RATE
in the phones as follows for
phones A and B.
A CODEC 1: G.711 u-law
A CODEC 2: NONE
A CODEC 3: NONE
B CODEC 1: G.711 u-law
B CODEC 2: NONE
B CODEC 3: NONE
PACKET RATE: 20ms
A calls B
Take a Wireshark trace of the
call.
Repeat the test for packet
rates of 10-40ms.
Expected Result
Codec and packet Rate will be
set as defined in the setup.
The Wireshark trace will show
that Codec will be G.711 u-law,
and the Packets will be 20ms
apart.
Comments
Phone A is the calling phone.
Phone B is the called phone.
To set the Codec and packet
Rate select: SETUP / INSTL /
KEYS / (select a primary line
key) / EDIT / CODEC
P/F
Packet rates displayed on the
Wireshark trace will reflect the
packet settings.
Set the CODEC/packet RATE Codec and packet Rate will be
in the phones as follows for
set as defined in the setup.
phones A and B.
The Wireshark trace will show
A CODEC 1: G.711 a-law
that Codec will be G.711 a-law,
A CODEC 2: NONE
and the Packets will be 20ms
A CODEC 3: NONE
apart.
B CODEC 1: G.711 a-law
B CODEC 2: NONE
B CODEC 3: NONE
PACKET RATE: 20ms
A calls B
Take a Wireshark trace of the
call.
Repeat the test for packet
Packet rates displayed on the
rates of 10-40ms.
Wireshark trace will reflect the
packet settings.
Set the CODEC/packet RATE Codec and packet Rate will be
in the phones as follows for
set as defined in the setup.
phones A and B.
The Wireshark trace will show
A CODEC 1: G.729
that Codec will be G.729, and
A CODEC 2: NONE
the Packets will be 20ms apart.
A CODEC 3: NONE
B CODEC 1: G.729
B CODEC 2: NONE
B CODEC 3: NONE
PACKET RATE: 20ms
A calls B
Take a Wireshark trace of the
call.
Repeat the test for packet
Packet rates displayed on the
rates of 10-40ms.
Wireshark trace will reflect the
packet settings.
Set the CODEC/packet RATE Codec and packet Rate will be
in the phones as follows for
set as defined in the setup.
PreReleasePhoneFeatureTest.doc
8/28/2014
pg. 11
Step
8
Action
phones A and B.
A CODEC 1: G.722
A CODEC 2: NONE
A CODEC 3: NONE
B CODEC 1: G.722
B CODEC 2: NONE
B CODEC 3: NONE
PACKET RATE: 20ms
A calls B
Take a Wireshark trace of the
call.
Repeat the test for packet
rates of 10-40ms.
9
Re-configure both phones
used, back to the default
settings.
Expected Result
The Wireshark trace will show
that Codec will be G.722, and
the Packets will be 20ms apart.
Comments
P/F
Packet rates displayed on the
Wireshark trace will reflect the
packet settings.
Phone default settings:
Codec 1=G.711 u-law
Codec 2=G.729
Codec 3=G722
Packet Rate=20ms
10
8.7 JITTER Test
8.7.1 JITTER, FIXED: INSTALL, KEYS, Edit Line Key
Step
Action
Expected Result
Select an existing line key,
The Line ID will blink red/green
and select EDIT / JTR
and the JITTER BUFFER
1
ADAPTIVE will appear, with
selections for DELAY and
FIXED.
Select FIXED.
The Line ID will blink red/green
and the JITTER
2
FIXED,DELAY=35 menu will
appear, with the selections:
DLY+, DLY-, and ADAPT
Adjust the jitter delay through The settings will all be
3
the settings to verify that they accepted.
are all accepted.
Re-configure the phone back
4
to its default settings
5
Comments
The default delay for FIXED
should be JITTER
FIXED,DELAY=35.
Fixed Jitter Range= 10-90
Default=35
Default = ADAPT
8.7.2 JITTER, ADAPTIVE: INSTALL KEYS, Edit Line Key
Step
Action
Expected Result
Comments
Select an existing line key,
The Line ID will blink red/green
and select EDIT / JTR /
and the JITTER BUFFER
1
ADAPT
ADAPTIVE will appear, with
selections for DELAY and
FIXED.
Select DELAY
The DELAY: MIN=10 MAX=100 The default Adaptive Jitter
display will appear with the
Delay range is 10-100ms.
2
selections: MIN+, MIN-, MAX+, Default=10
MAX-
PreReleasePhoneFeatureTest.doc
8/28/2014
P/F
P/F
pg. 12
Step
3
4
Action
Adjust the MIN and MAX
DELAY through their ranges
to verify that they are all
accepted.
Re-configure the phone back
to its default settings
Expected Result
Settings will all be accepted.
Comments
MIN range=0-280
MAX range=0-300
P/F
The default Adaptive Jitter
Delay range is 10-100ms.
Default=10
5
8.8 USER: RING, Personal Ringing
Test on 7810, 4104, 4101
Personal Settings: Tone, Offhook, Control, Local Call Forward (LCFWD)
LCFWD sets the timeout for Call Forward NO ANS
8.8.1 USER: RING, Personal Ringing, TONE
Step
Action
Expected Result
Check the default ring tone
Default tone=5
1
2
3
Try all of the ring tone
selections.
Call the phone at each ring
tone selection to verify that
the ring tones work when the
phone is called.
Comments
Setup / User / Ring / Tone / 1-6
After a phone Reset, the ring
tone default is 5.
There will be ring tones for keys When loaded with the xml file,
1-6, but not for 7-9
ring tone 8 loads and displays
as 8, but the phone rings at
tone 6.
The tones will reflect the
settings.
P/F
4
8.8.2
OFFHOOK Ringing
This controls the phone ringing when offhook. SINGLE or NORMAL, default=SINGLE
Step
Action
Expected Result
Comments
Check that the default setting
SETUP / USER / RING /
1
is SINGLE
OFFHOOK / SINGLE
Take the handset offhook,
The offhook phone should only
2
and then place a call to the
ring once, and should be at a
offhook phone.
reduced volume from normal.
3
Hang up the phones.
Change the setting to
SETUP / USER / RING /
4
NORMAL.
OFFHOOK / NORMAL
Take the handset offhook,
The offhook phone should ring
5
and then place a call to the
continuously, and be at a
offhook phone.
reduced volume from normal.
6
8.8.3
P/F
RING CONTROL
Individual line appearances can be set to always ring, never
ring, or ring after a delay. SETUP / USER / RING / CONTROL / (select a line key) / ALWAYS, NEVER,
WAIT2-7
The 4101 has no ring control; only test 7810 and 4104.
PreReleasePhoneFeatureTest.doc
8/28/2014
pg. 13
8.8.4 Always Ring
Step
Action
Set to Always Ring
Expected Result
1
Call the phone.
2
The phone should ring
continuously, and the line key
should blink green.
Comments
SETUP / USER / RING /
CONTROL / (select a line key) /
ALWAYS
Always Ring is the default.
On Teo UCM the call will get
transferred to voicemail after 4
rings.
P/F
3
8.8.5
Step
1
2
3
Never Ring
The 4101 has no ring control; only test 7810 and 4104.
Action
Expected Result
Comments
Set line 1 to NEVER ring and The phone will not ring, but the With a line set to NEVER ring
call the phone.
line key will light.
and USER / PREF set to
RINGING, with an incoming
call, when you pick up the
handset, the phone will select
an IDLE line, not the ringing
line, because the phone needs
to have ringing to pick up a
ringing line.
Make a call from line 1 to
another phone, and answer
the call.
Call the phone (line 2)
Line 2 will light, but no ring will On Teo UCM the call will get
be heard.
transferred to voicemail after 4
rings.
Ring after a Delay
The 4101 has no ring control; only test 7810 and 4104.
Step
Action
Expected Result
Comments
Check Delay 2 and Delay 7
On delay 2, the phone will not On Teo UCM the call will get
ring for the first 2 rings, and
transferred to voicemail after 4
then will begin ringing.
rings.
1
On delay 7, the call will be
transferred to voicemail before
the phone gets to the 7thring.
2
P/F
8.8.6
P/F
8.9 Adjusting the Ringer Volume
Test on 7810, 7810 + 8030X, 4104, 4101
The Volume keys adjust the ringer volume while you are not on a call.
The telephone will ring once with the new ringer volume setting, and
the new setting will be shown briefly in the display.
When the volume is at the lowest setting, the phone will display
"RINGER OFF" when idle.
Press the Volume Up key while you are not on a call to enable
the ringer.
PreReleasePhoneFeatureTest.doc
8/28/2014
pg. 14
Step
1
2
3
4
Action
Select the ring volume up
arrow repeatedly and verify
that the volume increases
with each press, until it
reaches the maximum
volume level.
Expected Result
Comments
Ring volume will increase each The beep lasts for about 2
time the volume up arrow is
seconds each time the volume
pressed, until the phone
up/down arrow is pressed.
reaches maximum volume.
Then the phone will still beep
when the up arrow is pressed,
but the volume will not increase
any more.
P/F
Call the phone and verify that
the phone rings with
maximum volume.
Select the ring volume down
arrow repeatedly and verify
that the volume decreases
with each press, until it
reaches the minimum volume
level (OFF).
Ring volume will decrease each
time the volume down arrow is
pressed, until the phone
reaches minimum volume (off).
Then the phone will continue to
not beep when the down arrow
is pressed.
Call the phone and verify that The phone will not ring, but the
the phone does not ring.
line key LED will still flash
green.
5
8.9.1
Distinctive Ringing of SIP Phone
Test on 7810, 4104, 4101
Tested on TEO-UC141 with an incoming pstn call, and an inside call.
Tested on Lucent for pre-emption
Step
Action
Expected Result
1
B goes offhook
B receives dialtone
A rings with inside ring tone, B
2
B dials A
has ringback
3
A goes offhook
A and B confirm talk path
4
B goes onhook
A in idle state
5
pstn dials A
A rings with outside ring tone
6
A goes offhook
A and pstn confirm talk path
7
C goes onhook
A in idle state
8
Comments
P/F
8.10 USER: DIR, User Directory
The User directory is available through the User/Dir menu selection, or by selecting the DIR key on the
keyboard.
Test on 7810, 7810 + 8030X, 4104, 4101
USER: DIR, Immediate or Editable Dialing
For one-touch dialing, enter a # character at the end of the dial string.
The string will be dialed immediately when the DIR key is selected. Without the #, when the directory
entry is selected, a line becomes active, and the number is dialed after the dial timeout expires. This
allows you to enter more digits after selecting the DIR entry.
Test on 7810, 7810 + 8030X, 4104, 4101.
Step
Action
Expected Result
Comments
P/F
1
Create call directory entries
To make a new DIR entry
with and without the # at the
select:
PreReleasePhoneFeatureTest.doc
8/28/2014
pg. 15
Step
Action
end of the number
Expected Result
Place calls with a number that
has the # at the end of the
number.
2
3
4
5
6
7
8
9
10
Comments
DIR / EDIT / (select an unused
key) / NUMBER / (enter a
number to dial) / OK / NAME /
(enter a name for the DIR entry)
/ OK. Then exit the DIR. Dial
Timeout is set at location
SETUP / INSTL / CALL / DIAL.
Default= 10 seconds.
There are 2 ways to dial from
the Directory.
1. pick up the handset first,
press the DIR key and then
select a dir entry.
2. Press the DIR key, select an
entry, and then pick up the
handset.
P/F
Verify the talk path and then
end the call.
Place calls with a number that You will have to wait for the dial
does not have the # at the
timeout to end, or press the
end of the number.
SEND softkey to complete
dialing the number.
Verify the talk path and then
end the call.
Create a Directory entry with The Dir entry will appear to be This is to enable the user to
a NUMBER and no NAME
blank, but will dial the number create hidden directory entries.
when selected.
Hidden directory entries always
sort to the end of the visible
directory.
Dial the hidden directory entry The number will dial.
Verify the talk path.
End the call.
8.11 Call Log
Stores up to 100 calls, first in, last out.
Us Asterisk for the load test, test, or an engineering server, do not use the house phone system. The test
runs from TCS Logview.
Run a load test to place calls into the call log of three phone types, 7810, 4104, and 4101.
A Phone RESTART clears the call log; the Error log must be cleared manually through the menus.
TO DUPLICATE:
Bring up 2 phones with no logs enabled. Connect to both phones through the Telnet; serial port can be
used. Place a call to a different phone than the ones used in the test, as a marker in the log, since all of
the other logs will be for the phone numbers used during the test.
At telnet port on the callee phone, enter the following:
CeTest
at1
ce1,100
At the telnet port on the caller phone, enter the following:
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CeTest
at1
cr1049#,100,100
Note "1049#" is the number of the target phone. The test will run for 100 calls.
If you need to end the test for some reason, press the Mute key on the phone. Disconnect Logview, clear
the Error Log, and Restart the phone to recover memory and clear the Call Log.
Test on 7810, 4104, 4101.
Step
Action
Expected Result
Comments
P/F
Restart a 4101, 4104, and a
The call logs will be cleared on Check the call log before and
1
7810
each phone
after the Restart to verify that
the log changed
Start a load test using the
Start the callee first. Then start
2
7810 as the callee, and the
the callers.
other two phones as callers.
make a call from each of the
callers to a phone not used in
3
the test, to be used as a
placeholder in the log.
Do the same for the callee.
Let the test run for at least
4
100 caller calls on each
caller.
Count the number of calls
The logs should display a
logged in each phone log.
message on both ends of the
5
log, stating NO NEWER
OUTGOING CALLS, OR NO
OLDER OUTGOING CALLS
6
8.12 HOT/WARM Dialing
Line Mode HOT/WARM/NORMAL
This is configured in the xml file as general settings.
Default=NORMAL
<line_mode>WARM</line_mode>
<hot_warm_username>1065</hot_warm_username>
<warmline_timeout>5</warmline_timeout>
Step
Action
Expected Result
A receives dialtone
1
A keys *96+4 digit #
C alerts to intercom
A and C confirm talk path
C goes onhook
A in idle state
2
3
4
Comments
A must have Intercom COS bit
set
xxxx is extension of phone C
that supports intercom, or has
Voice Announce enabled.
P/F
Comments
SETUP / ADMIN / INSPCT
P/F
8.13 ADMIN: INSPCT, Inspect Keys
Step
1
Test on 7810, 7810 + 8030X, 4104, 4101
Action
Expected Result
Inspect all key types.
Types should all be reported
correctly
2
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8.14 INSTALL: PASSWD, Installation Password
Password was change to PIN with version 0x.03.15.15.
<install_pin></install_pin>
The PIN is defined as 4 to 20 numbers, no alphabet characters or special characters.
After the PIN is entered one time, it has to be entered every time after that.
The PIN can only be deleted by Resetting the phone, or entering a null pin with the xml tag.
Test on 7810, 7810 + 8030X, 4104, 4101
Step
Action
Expected Result
Comments
Using the menu, set a PIN of
1
4 numbers and verify that it
works.
Using the menu, set a PIN of
2
20 numbers and verify that it
works.
Test the boundaries.
5 and 21 numbers, Alpha
3
characters, special characters,
null.
Verify that once you have put
The only way to clear the PIN is
4
in a PIN, that you have to use
to Reset the phone, or enter a
a PIN after that.
PIN of null with the xml file.
Use the xml file to null out the The install pin should be
5
install pin.
removed.
6
P/F
8.15 USER: PIN, Setting a Call Log PIN
Test on 7810, 4104, 4101
Original Password Requirements
Passwords must be at least 9 characters long, and must
contain at least :
2 digits
2 upper case letters
2 lower case letters
2 special characters
The password cannot contain the line ID, forward or backward.
At least 4 characters must have changed from previous password entries.
Select HELP to show these password requirements on screen.
Press the OK key.
Repeat the password when prompted to verify the new entry.
Press the OK key.
“PASSWORD SET” will be displayed to confirm the new password.
Record your password for future reference.
Press the OK key to return to the User Options menu or press
the SETUP key to exit Setup Mode.
The password requirements are now limited to a range of 4 to 20 numbers for all phone types, and there
is no Help screen.
Step
Action
Expected Result
Comments
P/F
1
Set PIN for the User Log.
Use the PIN to verify that they
2
were set.
3
Clear the PIN.
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Step
4
5
6
7
Action
Enter the Log to verify that
the passwords were cleared
Try alpha characters and
special characters.
Check the limits of the PIN.
Expected Result
Comments
P/F
Only numbers will be allowed.
4-20 numbers
8.16 INSTALL: IP, IP Addresses, Syslog
In order to test the Syslog function, use your PC address as the Syslog server address, and run the
TFTPD32 program on the PC, with Syslog enabled.
<ip_syslog source=”DHCP4”>192.168.72.133</ip_syslog>
Logging option for the Syslog Server operation.
<syslog_option>NONE</syslog_option>
Valid options are,
NONE – Disabled / No Logging (Default)
BASIC – SIP Phone Error Logs are sent to Syslog server
QOS – Error Logs and Quality of Service call packet statistics sent.
In the 4101, you can only set the Syslog IP address in the phone menu, and the Option defaults to NONE.
To set the Syslog Option to anything besides NONE, you have to set the Syslog Option in the xml file,
using the tag:
<syslog_option>NONE</syslog_option>
NONE – Disabled / No Logging (Default)
BASIC – SIP Phone Error Logs are sent to Syslog server
QOS – Error Logs and Quality of Service call packet statistics sent.
Step
Action
Expected Result
Comments
P/F
Set up a Syslog server on the
Run the TFTPD32 program on
1
Test PC
the PC, with the Syslog option
enabled.
Set the Syslog Option menu
Verify that no logs are sent to
selection to NONE.
the Syslog server when the
2
phone is restarted, or when
calls are made.
Set the Syslog Option menu
Verify that SIP Phone Error logs
3
selection to BASIC.
re sent to the Syslog server.
Set the Syslog Option menu
Verify that Error Logs and
selection to QOS.
Quality of Service call packet
4
statistics are sent.
5
9 ISSUES:
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