prerelease phone feature test

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 featuresThis version is for use on Teo UCM, Avaya, Redcom, BroadSoft, Cisco, NECTABLE OF CONTENTSCONTENTPAGE NUMBER21Scope

22Hardware used for the test

23Test Servers

24Software versions used during the test

25Test Responses

36Test Failures

37Order of Test

38Regression Tests

38.1INSTALL: RESET, MAC xml

48.2INSTALL: RESET, LINE xml

48.3PROGRAM UPDATE

48.3.1INSTALL: UPDATE, PRGRM, TFTP Program Update

58.3.2INSTALL: UPDATE, PRGRM, Program Update Not Required

58.3.3INSTALL: UPDATE, PRGRM, HTTP or HTTPS Program Update

68.3.4INSTALL: UPDATE, PRGRM, TEO

78.4USER: VOICE, VOLUME

78.4.1USER: VOICE, VOLUME, Receive

78.4.2USER: VOICE, VOLUME, Transmit

78.4.3USER: VOICE, VOLUME, Reset to Defaults

78.5INSTALL: QOS, Quality of Service

98.6CODEC TESTING

98.6.1INSTALL: Keys, Edit Line Key, CODEC

118.6.2AUTO CODEC SELECTION: Install, Keys, Edit Line Key

128.7JITTER Test

128.7.1JITTER, FIXED: INSTALL, KEYS, Edit Line Key

138.7.2JITTER, ADAPTIVE: INSTALL KEYS, Edit Line Key

138.8USER: RING, Personal Ringing

138.8.1USER: RING, Personal Ringing, TONE

138.8.2OFFHOOK Ringing

148.8.3RING CONTROL

148.8.4Always Ring

148.8.5Never Ring

148.8.6Ring after a Delay

158.9Adjusting the Ringer Volume

158.9.1Distinctive Ringing of SIP Phone

168.10USER: DIR, User Directory

178.11Call Log

188.12HOT/WARM Dialing

188.13ADMIN: INSPCT, Inspect Keys

188.14INSTALL: PASSWD, Installation Password

188.15USER: PIN, Setting a Call Log PIN

198.16INSTALL: IP, IP Addresses, Syslog

209ISSUES:

1 ScopeAn 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 test78107810-TSG7810 + 8030X410441013 Test ServersTeo UCM4 Software versions used during the testPhone Software: v0x.04.205 Test Responses, P: Pass, F: FailN/T: Not TestedN/A: Not ApplicableBUG: Issues that occurred during the test.6 Test FailuresTest 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 TestThis list is provided as a logical progression of events and tests to be performed.TestTime EstimatePASS/FAIL

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

8 Regression Tests8.1 INSTALL: RESET, MAC xml

Test on 7810, 7810 + 8030X, 4104, 4101

StepActionExpected ResultCommentsP/F

1 Configure the TCS7000A.xml file (global xml tag file) with a configuration tag set for MAC address.Set the TCS7000A config tag to MAC: MAC

Or comment out the line; MAC is the default

2 Create a unique configuration xml file for the test phone, with the file name made up of the phone MAC address.i.e. 00048DFFFFE8.xml

3 Reset the phone, and watch the TFTP server windowThe phone will load the TCS7000A.xml file and its MAC.xml file.

The phone will initialize and register.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

StepActionExpected ResultCommentsP/F

1 Configure the TCS7000A.xml file (global xml tag file) with a configuration tag set for LINE address.Set the TCS7000A config tag to LINE:

LINE

2 Create a unique configuration xml file for the test phone, with the file name made up of the phone Line ID.

3 Reset the phone, and watch the TFTP server window.The phone will load the TCS7000A.xml file and its Line ID.xml file.

The phone will initialize and register.As the phone initializes, it will prompt the user for needed data.

4

8.3 PROGRAM UPDATE8.3.1 INSTALL: UPDATE, PRGRM, TFTP Program UpdateTest on 7810, 7810 + 8030X, 4104, 4101StepActionExpected ResultCommentsP/F

1 Provision the Update Server with a different version of software from the current version running in the UUT.

2 Configure the UUT for TFTP updateSETUP / INSTL / (right arrow) / UPDATE / PRGRM / PROTO / TFTP

3 Select "UPDATE / PRGRM"to load the different software from the Update Server.The UUT should load the different software, and RESTART. After RESTART the phone should successfully register, and have the same LINE ID and other settings as before the update.SETUP / INSTL / (right arrow) / UPDATE / PRGRM / START

4 Check the error log for problems.There should be no unexpected errors in the log.SETUP / INSTL / (right arrow) / (right arrow) / LOG / ERROR

5 Check the phone S/W version to see that it matches the version just loadedIt should be the new version.SETUP / ADMIN / VERS / S/W

6

8.3.2 INSTALL: UPDATE, PRGRM, Program Update Not RequiredTest on 7810, 7810 + 8030X, 4104, 4101StepActionExpected ResultCommentsP/F

1 Provision the Update Server with the same version of software that is presently running in the phone.SETUP / ADMIN / VERS / S/W

2 Configure the UUT for TFTP updateSETUP / INSTL / (right arrow) / UPDATE / PRGRM / PROTO / TFTP

3 Perform a Update Program to attempt to load the same software from the Update Server.The UUT should not load the software into the phone. The phone will display:

PROGRAM UPDATE NOT REQUIREDSETUP / INSTL / (right arrow) / UPDATE / PRGRM / START


5

8.3.3 INSTALL: UPDATE, PRGRM, HTTP or HTTPS Program UpdateTest on 7810, 7810 + 8030X, 4104, 4101StepActionExpected ResultCommentsP/F

1 Create a location in the computer Update server path, for program update files. This will be a different location than for TFTP updates.

This path is where the digitally signed files are located under the update server root directory.

Place the folder containing the phone update program files in the HTTP/HTTPS update file address. Also place all of the folder contents in the same location.

Or do a Program Update from the keypad on a phone connected to a TEO UCM server.Server name = phonemgr-w2k8.greyhawk.tonecommander.com

Server name with file location appended = phonemgr-w2k8.greyhawk.tonecommander.com/teotestdata/stever.This is the location that the HTTPS server is pointed at.If you are connected to a TEO UCM server, Program Updates are done with HTTPS protocol. This is adequate for this test.

2 Edit the phone's UPDATE IP address (Domain Name), appending the location for the 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.

The program update file location that was just created.

Example:

Server name = phonemgr-w2k8.greyhawk.com

Or 192.168.72.60

Server name with file location appended = phonemgr-w2k8.greyhawk.com/teo-data/stever

or 192.168.72.60/teo-data/stever

(wwwroot is not needed in the domain name)

3 Enter the following key sequence on the phone to set the phones update protocol to HTTP or HTTPS:

On the phone, select SETUP / INSTL / (right arrow) / UPDATE / PRGRM / PROTO / HTTP(or HTTPS)

Sets the phone update protocol to HTTP or HTTPS

4 Provision the Update Server with a different version of software from the current version running in the UUT.You will be using the HTTP file location in the server for the update.

5 To perform the phone program update select the following phone key sequence:

SETUP / INSTL / (right arrow) / UPDATE / PRGRM / STARTThe 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.


7 Check the phone S/W version to see that it matches the version just loadedIt should be the sameSETUP / ADMIN / VERS / S/W.

Or select the INFO key on the phone.

8

8.3.4 INSTALL: UPDATE, PRGRM, TEOTest on 7810, 7810 + 8030X, 4104, 4101StepActionExpected ResultCommentsP/F

1 Change the Update server address to be the same as the UCM server.

2 Configure the phone for TEO update protocol.The phone will require a restart. Select YES when prompted.SETUP / INSTL / UPDATE / PROGRAM / PROTO / TEO

3 Perform a Update Program SETUP / INSTL / (right arrow) / UPDATE / PRGRM / START

4 Check the error log for problems after the update is complete.There should be no unexpected errors in the log.SETUP / INSTL / (right arrow) / (right arrow) / LOG / ERROR

5

8.4 USER: VOICE, VOLUMETest on 7810, 7810 + 8030X, 4104, 41018.4.1 USER: VOICE, VOLUME, ReceiveThe 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.StepActionExpected ResultCommentsP/F

1 Check the limits of the volume for functionality.The menu should work as expected.The volume settings should not distort the audio, and the volume should be reasonably usable.SETUP / USER / VOICE / VOLUME / (HAND, or HEAD)

2

8.4.2 USER: VOICE, VOLUME, TransmitStepActionExpected ResultCommentsP/F

1 Check the limits of the Transmit volume for functionality.The menu should work as expected.

The volume settings should not distort the audio, and the volume should be reasonably usable.SETUP / USER / VOICE / VOLUME / (HAND, HEAD, or SPKR)

2

8.4.3 USER: VOICE, VOLUME, Reset to DefaultsStepActionExpected ResultCommentsP/F

1 Set the volume levels to values other than default, and select the Reset keyLevels should all be reset to level 4.SETUP / USER / VOICE / VOLUME / RESET

2

8.5 INSTALL: QOS, Quality of ServiceQoS uses diffserv valuesDifferentiated 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 low-latency 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:Default PHB (Per hop behavior)which is typically best-effort trafficExpedited Forwarding (EF) PHBdedicated to low-loss, low-latency trafficAssured Forwarding (AF) PHBgives assurance of delivery under prescribed conditionsClass Selector PHBswhich 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 LevelPriority PrefixHex numberDecimal numberComment

00

2840

Override902941

Flash912b43

Immediate922d45

2e46Default QoS

Priority932f47

Routine943149

3f63

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 decimalQoS 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 SettingQoS Signal SettingPriority PrefixInvite DSCPRTP DSCP

4640 (28 hex)None2e2e

4140 (28 hex)90, Override2929 (Override)

4140 (28 hex)90, Override2929 (Override)

4340 (28 hex)91, Flash2b2b (Flash)

4540 (28 hex)92, Immediate2d2d (Immediate)

4740 (28 hex)93, Priority2f2f (Priority)

4940 (28 hex)94, Routine3131 (Routine)

Test on 7810, 7810 + 8030X, 4104Range=0-63 (decimal) default: voice=49, signal=40The entire DSCP number appears as 0x2e, which is a 6 bit number; third and 4th values are the DSCP value (hex).StepActionExpected ResultCommentsP/F

1 Verify that the QoS settings are set at default.Default:Voice=49

Signal=40SETUP / INSTL / QoS / L3 / VOICE or SIGNAL

2 Run Wireshark

3 Call between two phones, then hang up the call

4 Stop WiresharkQoS 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

5 Change the values to 0 and retest

6 Change the values to 63 and retestThe DSCP value will be 3f(hex).

7

8.6 CODEC TESTING8.6.1 INSTALL: Keys, Edit Line Key, CODECServer in bypass mode. This test tests Codec functionality.The CODEC menu is used to set the phones 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.Admin/Diag/Packet/Active, during a call to confirm codec settings through the menu.Wireshark trace.Test on 7810, 4104, 4101.StepActionExpected ResultCommentsP/F

1 On the UUT select SETUP / INSTALL / KEYS / (select the primary line key) / EDIT / CODEC, and set CODEC 1 to each of the CODEC settings in the calling phone, and packet RATES, one at a time and perform the following test steps.The Line ID will blink red/green and the 1=G711 u-LAW menu will appear, with the selection for CODEC.Select CODECS G711, G729, and G722, one at a time. Set CODECS 2 and 3 to NONE for each test. Also set packet RATES 10, 20, 30, and 40ms for each CODEC.

Use these test settings for CODEC 1 and corresponding packet RATE:

G.711 u-LAW, G.711 a-LAW, G.729, G722 with Packet settings 10-40ms at each setting.

2 Make a call between the UUT and another phone while monitoring a Wireshark trace.Both phones will use the same CODEC (the one selected as CODEC 1 on the calling phone), and the phones will have a voice path.

The CODECS will appear in the call INVITE to the server.

The time between packets, for 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.The CODECS will appear in the correct order in the call INVITE.

RTP packets from both phones will be using the same CODEC.

Test all of the settings as described in the setup in the first step.

3 Re-configure both phones used, back to the default settings.Default Codec settings are:

1=G711 u-LAW

2=G729

3=G722

Packet rate=20ms

4

8.6.2 AUTO CODEC SELECTION: Install, Keys, Edit Line KeyServer 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.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.Test on 7810, 4104, 4101StepActionExpected ResultCommentsP/F

1 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.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.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

2 Repeat the test for packet rates of 10-40ms.Packet rates displayed on the Wireshark trace will reflect the packet settings.


A CODEC 1: G.711 a-law

A CODEC 2: NONE

A CODEC 3: NONE

B CODEC 1: G.711 a-law

B CODEC 2: NONE

B CODEC 3: NONE

PACKET RATE: 20ms

A calls B


The Wireshark trace will show that Codec will be G.711 a-law, and the Packets will be 20ms apart.



A CODEC 1: G.729

A CODEC 2: NONE

A CODEC 3: NONE

B CODEC 1: G.729

B CODEC 2: NONE

B CODEC 3: NONE

PACKET RATE: 20ms

A calls B


The Wireshark trace will show that Codec will be G.729, and the Packets will be 20ms apart.



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


The Wireshark trace will show that Codec will be G.722, and the Packets will be 20ms apart.


9 Re-configure both phones used, back to the default settings.Phone default settings:

Codec 1=G.711 u-law

Codec 2=G.729

Codec 3=G722

Packet Rate=20ms

10

8.7 JITTER Test8.7.1 JITTER, FIXED: INSTALL, KEYS, Edit Line KeyStepActionExpected ResultCommentsP/F

1 Select an existing line key, and select EDIT / JTRThe Line ID will blink red/green and the JITTER BUFFER ADAPTIVE will appear, with selections for DELAY and FIXED.

2 Select FIXED.The Line ID will blink red/green and the JITTER FIXED,DELAY=35 menu will appear, with the selections: DLY+, DLY-, and ADAPTThe default delay for FIXED should be JITTER FIXED,DELAY=35.

3 Adjust the jitter delay through the settings to verify that they are all accepted.The settings will all be accepted.Fixed Jitter Range= 10-90

Default=35

4 Re-configure the phone back to its default settingsDefault = ADAPT

5

8.7.2 JITTER, ADAPTIVE: INSTALL KEYS, Edit Line KeyStepActionExpected ResultCommentsP/F

1 Select an existing line key, and select EDIT / JTR / ADAPTThe Line ID will blink red/green and the JITTER BUFFER ADAPTIVE will appear, with selections for DELAY and FIXED.

2 Select DELAYThe DELAY: MIN=10 MAX=100 display will appear with the selections: MIN+, MIN-, MAX+, MAX-The default Adaptive Jitter Delay range is 10-100ms.

Default=10

3 Adjust the MIN and MAX DELAY through their ranges to verify that they are all accepted.Settings will all be accepted.MIN range=0-280

MAX range=0-300

4 Re-configure the phone back to its default settingsThe default Adaptive Jitter Delay range is 10-100ms.

Default=10

5

8.8 USER: RING, Personal RingingTest on 7810, 4104, 4101Personal Settings: Tone, Offhook, Control, Local Call Forward (LCFWD)LCFWD sets the timeout for Call Forward NO ANS8.8.1 USER: RING, Personal Ringing, TONEStepActionExpected ResultCommentsP/F

1 Check the default ring toneDefault tone=5Setup / User / Ring / Tone / 1-6

After a phone Reset, the ring tone default is 5.

2 Try all of the ring tone selections.There will be ring tones for keys 1-6, but not for 7-9When loaded with the xml file, ring tone 8 loads and displays as 8, but the phone rings at tone 6.

3 Call the phone at each ring tone selection to verify that the ring tones work when the phone is called.The tones will reflect the settings.

4

8.8.2 OFFHOOK RingingThis controls the phone ringing when offhook. SINGLE or NORMAL, default=SINGLEStepActionExpected ResultCommentsP/F

1 Check that the default setting is SINGLESETUP / USER / RING / OFFHOOK / SINGLE

2 Take the handset offhook, and then place a call to the offhook phone.The offhook phone should only ring once, and should be at a reduced volume from normal.

3 Hang up the phones.

4 Change the setting to NORMAL.SETUP / USER / RING / OFFHOOK / NORMAL

5 Take the handset offhook, and then place a call to the offhook phone.The offhook phone should ring continuously, and be at a reduced volume from normal.

6

8.8.3 RING CONTROLIndividual line appearances can be set to always ring, neverring, or ring after a delay. SETUP / USER / RING / CONTROL / (select a line key) / ALWAYS, NEVER, WAIT2-7The 4101 has no ring control; only test 7810 and 4104.8.8.4 Always RingStepActionExpected ResultCommentsP/F

1 Set to Always RingSETUP / USER / RING / CONTROL / (select a line key) / ALWAYS

Always Ring is the default.

2 Call the phone.The phone should ring continuously, and the line key should blink green.On Teo UCM the call will get transferred to voicemail after 4 rings.

3

8.8.5 Never RingThe 4101 has no ring control; only test 7810 and 4104.StepActionExpected ResultCommentsP/F

1 Set line 1 to NEVER ring and call the phone.The phone will not ring, but the line key will light.With a line set to NEVER ring 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.

2 Make a call from line 1 to another phone, and answer the call.

3 Call the phone (line 2)Line 2 will light, but no ring will be heard.On Teo UCM the call will get transferred to voicemail after 4 rings.

1

8.8.6 Ring after a DelayThe 4101 has no ring control; only test 7810 and 4104.StepActionExpected ResultCommentsP/F

1 Check Delay 2 and Delay 7On delay 2, the phone will not ring for the first 2 rings, and then will begin ringing.

On delay 7, the call will be transferred to voicemail before the phone gets to the 7thring.On Teo UCM the call will get transferred to voicemail after 4 rings.

2

8.9 Adjusting the Ringer VolumeTest on 7810, 7810 + 8030X, 4104, 4101The Volume keys adjust the ringer volume while you are not on a call.The telephone will ring once with the new ringer volume setting, andthe 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 enablethe ringer.StepActionExpected ResultCommentsP/F

1 Select the ring volume up arrow repeatedly and verify that the volume increases with each press, until it reaches the maximum volume level.Ring volume will increase each time the volume up arrow is pressed, until the phone reaches maximum volume. Then the phone will still beep when the up arrow is pressed, but the volume will not increase any more.The beep lasts for about 2 seconds each time the volume up/down arrow is pressed.

2 Call the phone and verify that the phone rings with maximum volume.

3 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.

4 Call the phone and verify that the phone does not ring.The phone will not ring, but the line key LED will still flash green.

5

8.9.1 Distinctive Ringing of SIP PhoneTest on 7810, 4104, 4101Tested on TEO-UC141 with an incoming pstn call, and an inside call.Tested on Lucent for pre-emptionStepActionExpected ResultCommentsP/F

1 B goes offhookB receives dialtone

2 B dials AA rings with inside ring tone, B has ringback

3 A goes offhookA and B confirm talk path

4 B goes onhookA in idle state

5 pstn dials AA rings with outside ring tone

6 A goes offhookA and pstn confirm talk path

7 C goes onhookA in idle state

8

8.10 USER: DIR, User DirectoryThe 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, 4101USER: DIR, Immediate or Editable DialingFor 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.StepActionExpected ResultCommentsP/F

1 Create call directory entries with and without the # at the end of the numberTo make a new DIR entry select:

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.

2 Place calls with a number that has the # at the end of the number.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.

3 Verify the talk path and then end the call.

4 Place calls with a number that does not have the # at the end of the number.You will have to wait for the dial timeout to end, or press the SEND softkey to complete dialing the number.

5 Verify the talk path and then end the call.

6 Create a Directory entry with a NUMBER and no NAMEThe Dir entry will appear to be blank, but will dial the number when selected.This is to enable the user to create hidden directory entries. Hidden directory entries always sort to the end of the visible directory.

7 Dial the hidden directory entryThe number will dial.

8 Verify the talk path.

9 End the call.

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8.11 Call LogStores 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:CeTestat1ce1,100At the telnet port on the caller phone, enter the following:CeTestat1cr1049#,100,100Note "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.StepActionExpected ResultCommentsP/F

1 Restart a 4101, 4104, and a 7810The call logs will be cleared on each phoneCheck the call log before and after the Restart to verify that the log changed

2 Start a load test using the 7810 as the callee, and the other two phones as callers.Start the callee first. Then start the callers.

3 make a call from each of the callers to a phone not used in the test, to be used as a placeholder in the log.

Do the same for the callee.

4 Let the test run for at least 100 caller calls on each caller.

5 Count the number of calls logged in each phone log.The logs should display a message on both ends of the log, stating NO NEWER OUTGOING CALLS, OR NO OLDER OUTGOING CALLS

6

8.12 HOT/WARM DialingLine Mode HOT/WARM/NORMALThis is configured in the xml file as general settings.Default=NORMAL

WARM

1065

5StepActionExpected ResultCommentsP/F

1 A receives dialtoneA must have Intercom COS bit set

2 A keys *96+4 digit #C alerts to intercom

A and C confirm talk pathxxxx is extension of phone C that supports intercom, or has Voice Announce enabled.

3 C goes onhookA in idle state

4

8.13 ADMIN: INSPCT, Inspect KeysTest on 7810, 7810 + 8030X, 4104, 4101StepActionExpected ResultCommentsP/F

1 Inspect all key types.Types should all be reported correctlySETUP / ADMIN / INSPCT

2

8.14 INSTALL: PASSWD, Installation PasswordPassword was change to PIN with version 0x.03.15.15.

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, 4101StepActionExpected ResultCommentsP/F

1 Using the menu, set a PIN of 4 numbers and verify that it works.

2 Using the menu, set a PIN of 20 numbers and verify that it works.

3 Test the boundaries.5 and 21 numbers, Alpha characters, special characters, null.

4 Verify that once you have put in a PIN, that you have to use a PIN after that.The only way to clear the PIN is to Reset the phone, or enter a PIN of null with the xml file.

5 Use the xml file to null out the install pin.The install pin should be removed.

6

8.15 USER: PIN, Setting a Call Log PINTest on 7810, 4104, 4101Original Password RequirementsPasswords must be at least 9 characters long, and mustcontain at least :2 digits2 upper case letters2 lower case letters2 special charactersThe 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 pressthe 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.StepActionExpected ResultCommentsP/F

1 Set PIN for the User Log.

2 Use the PIN to verify that they were set.

3 Clear the PIN.

4 Enter the Log to verify that the passwords were cleared

5 Try alpha characters and special characters.Only numbers will be allowed.

6 Check the limits of the PIN.4-20 numbers

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8.16 INSTALL: IP, IP Addresses, SyslogIn 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.192.168.72.133Logging option for the Syslog Server operation.NONEValid options are,NONE Disabled / No Logging (Default)BASIC SIP Phone Error Logs are sent to Syslog serverQOS 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:NONENONE Disabled / No Logging (Default)BASIC SIP Phone Error Logs are sent to Syslog serverQOS Error Logs and Quality of Service call packet statistics sent.StepActionExpected ResultCommentsP/F

1 Set up a Syslog server on the Test PCRun the TFTPD32 program on the PC, with the Syslog option enabled.

2 Set the Syslog Option menu selection to NONE.Verify that no logs are sent to the Syslog server when the phone is restarted, or when calls are made.

3 Set the Syslog Option menu selection to BASIC.Verify that SIP Phone Error logs re sent to the Syslog server.

4 Set the Syslog Option menu selection to QOS.Verify that Error Logs and Quality of Service call packet statistics are sent.

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9 ISSUES:PreReleasePhoneFeatureTest.doc8/28/2014pg. 9

prerelease phone feature test

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