Page History

1001

Verify SIP device registers through SBC 

Pass

1002

Verify SIP device registers through SBC with authentication

Pass

2001

PSTN calls SIP device, PSTN hangs up 

Pass

2002

PSTN calls SIP device, SIP hangs up 

Pass

2003

PSTN calls SIP device, PSTN hangs up before call connects 

Pass

2004

PSTN calls SIP device, SIP device not registered 

Pass

2005

PSTN call SIP device, SIP busy

Pass

2006

PSTN call SIP device, SIP no answer

Pass

2007

3 PSTNs call SIP device simultaneously

Pass

3001

SIP calls PSTN, PSTN hangs up

Pass

3002

SIP calls PSTN, SIP hangs up

Pass

3003

SIP calls PSTN, SIP hangs up before call connects

Pass

3004

SIP calls PSTN, Privacy enabled

Pass

3005

SIP calls PSTN, PSTN does not exists 

Pass

3006

SIP calls PSTN, no answer

Pass

4001

ATTENDED SIP #1 calls PSTN, SIP #1 transfers PSTN to SIP #2 -reinvite

Not Applicable

4002

ATTENDED SIP #1 calls PSTN #1, SIP #1 transfers PSTN #2 -reinvite

Not Applicable

4003

ATTENDED PSTN calls SIP #1, SIP #1 transfer PSTN to SIP #2 -reinvite

Not Applicable

4004

 ATTENDED PSTN #1 calls SIP #1, SIP #1 transfer PSTN #1 to PSTN #2 -reinvite

Not Applicable

4005

 BLIND SIP #1 calls PSTN, SIP #1 transfers PSTN to SIP #2 -reinvite

Not Applicable

4006

 BLIND SIP #1 calls PSTN #1, SIP #1 transfers PSTN #2 -reinvite

Not Applicable

4007

 BLIND PSTN calls SIP #1, SIP #1 transfer PSTN to SIP #2 -reinvite

Not Applicable

4008

 BLIND PSTN #1 calls SIP #1, SIP #1 transfer PSTN #1 to PSTN #2 -reinvite

Not Applicable

4009

ATTENDED SIP #1 calls PSTN, SIP #1 transfers PSTN to SIP #2 - refer

Pass

4010

ATTENDED SIP #1 calls PSTN #1, SIP #1 transfers PSTN #2 - refer

Pass

4011

ATTENDED PSTN calls SIP #1, SIP #1 transfer PSTN to SIP #2 - refer

Pass

4012

ATTENDED PSTN #1 calls SIP #1, SIP #1 transfer PSTN #1 to PSTN #2 - refer

Pass

4013

BLIND SIP #1 calls PSTN, SIP #1 transfers PSTN to SIP #2 - refer

Pass

4014

BLIND SIP #1 calls PSTN #1, SIP #1 transfers PSTN #2 - refer

Pass

4015

BLIND PSTN calls SIP #1, SIP #1 transfer PSTN to SIP #2 - refer

Pass

4016

BLIND PSTN #1 calls SIP #1, SIP #1 transfer PSTN #1 to PSTN #2 - refer

Pass

5001

SIP #1 calls PSTN, SIP #1 conferences in SIP #2

Pass

5002

SIP #1 calls PSTN #1, SIP #1 conferences in PSTN #2 

Pass

5003

PSTN calls SIP #1, SIP #1 conferenecs in SIP #2

Pass

5004

PSTN #1 SIP #1, SIP #1 conferenecs in PSTN #2

Pass

5005

PSTN, SIP #1, and SIP #2 call into conferencing bridge

Pass

6001

SIP #1 sets call forwarding to SIP #2 - reinvite

Pass

6002

SIP #1 sets call forwarding to PSTN - reinvite

Pass

6003

SIP #1 sets call forwarding to SIP #2 -refer

Not Applicable

6004

SIP #1 sets call forwarding to PSTN - refer

Not Applicable

6005

SIP #1 sets call forwarding to SIP #2 - 302 Moved Temporarily 

Not Applicable

6006

SIP #1 sets call forwarding to PSTN - 302 Moved Temporarily 

Not Applicable

6007

SIP #1 set call forwarding on BUSY

Pass

6008

SIP #1 set call forwarding on ring no answer

Pass

7001

SIP calls n11 (e.g. 211)

Pass

7002

SIP calls 911

Pass

7003

SIP calls 0+Local PSTN (operator assisted)

Pass

7004

SIP calls 0 (operator calls)

Pass

7005

SIP dials 011+18 digits (international number)

Pass

7006

SIP dials 1+10 digits (long distance)

Pass

7007

SIP dial 1-800+7 digits (toll free number) 

Pass

8001

SIP calls PSTN, SIP puts PSTN on hold then resumes call

Pass

8002

SIP calls PSTN, SIP puts PSTN on hold for a long time (10min)

then resumes call

Pass

8003

SIP calls PSTN, SIP puts PSTN on hold then resumes call - with music

Pass

8004

SIP calls PSTN, SIP puts PSTN on hold for a long time (10 min)

then resumes call - with music

Pass

9001

SIP G.711 calls PSTN (IVR).

Navigates through IVR menu tree successfully - inband

Conditional Pass

9002

SIP G.729 calls PSTN (IVR).

Navigates through IVR menu tree successfully - inband

Conditional Pass

9003

SIP G.711 calls PSTN (IVR).

Navigates through IVR menu tree successfully -RFC 2833 out of band

Conditional Pass

9004

SIP G.729 calls PSTN (IVR).

Navigates through IVR menu tree successfully -RFC 2833 out of band

Conditional Pass

9005

PSTN to SIP. No answer. Leave voicemail

Not Applicable

9006

SIP retrieves voicemail

Not Applicable

10001

set G.711 prefered, SIP calls PSTN

Pass

10002

set G.729 prefered SIP calls PSTN

Pass

11001

PSTN to SIP G.711 fax

Not Applicable

11002

SIP to PSTN G.711 fax

Not Applicable

11003

PSTN to SIP T.38 fax

Not Applicable

11004

SIP to PSTN T.38 fax

Not Applicable

12001

PSTN calls SIP. Call last for at least 25 minutes

Pass

13001

Inbound call 

PSTN calls SIP device, PSTN hangs up 

Pass

Self-signed certificates were used on this test.

13002

Inbound call 

PSTN calls SIP device, SIP hangs up 

Pass

13003

Outbound call 

SIP calls PSTN, PSTN hangs up

Pass

13004

Outbound call 

SIP calls PSTN, SIP hangs up

Pass

14001

Inbound call 

PSTN calls SIP device, PSTN hangs up 

Pass

14002

Inbound call 

PSTN calls SIP device, SIP hangs up 

Pass

14003

Outbound call 

SIP calls PSTN, PSTN hangs up

Pass

14004

Outbound call 

SIP calls PSTN, SIP hangs up

Pass

Setup:

1. A, B, C are register

2. On the phone, configure for conference type = Network Conference

Steps:

1. A call B

2. B answer call. Verify speech path

3. A make new call to C

4. C answer the call

5. A merge B &C into conference

Setup:

1. 4-10 users register

2. Configure(Gencom) for conference type = Network Conference

Steps:

1. U1 call U2

2. U2 answer call. Verify speech path

3. U1 make new call to U3

4. U3 answer the call

5. U1 merge U2 & U3 into conference

6. U1 continue making conference with U4 (up to U10)

Setup:
1. A, B register

Steps:
1. A call B with blocking CLI using VSC code + B's DN (*67 + DN).

Setup:
1. A, B register
2. From A's EUP, enable ID Restriction

Steps:
1. A enable CLI when calling to B using VSC code + B's DN (*39 + DN).

Setup:
Subscriber A and B are registered and both assigned Multimedia Office Premium

Steps:
1. PSTN calls subscriber A
2. Subscriber A, puts call on hold and dials <Call_Park_Directed VSC> + <B's VoIP Number>
3. Subscriber B dials <Call_Retrieve_Directed VSC>

Setup:
Subscriber A and B are registered and both assigned Multimedia Office Premium

Steps:
1. PSTN calls subscriber A
2. Subscriber A, puts the call on hold and dials <Call_Park_General_Parking VSC>
3. Subscriber B dials <Call_Retrieve_General_Parking VSC> + <Parking Lot Number>

Setup:
Add a pickup group including A, B
A, B, C register

Steps:
1. C call A
2. B pickup the call using pickupDN: VSC code + group's DN (*31 + group DN)

Setup:
Add a pickup group including A, B
A, B, C register

Steps:
1. C call A
2. B pickup the call using pickupTarget: VSC code + A's DN (*31 + A's DN)

Setup:
Add a pickup group including A, B
A, B, C register

Steps:
1. C call A
2. B pickup the call using pickupDirected: VSC code (*31)

Setup:
1. A, B, C register
2. From B's EUP, select the Routing tab, add an advance route:
When a call is received, From these numbers -> enter A's URI
Route: Ring these numbers: enter C's URI

Steps:
1. A call B.

Setup:
1. A, B, C, D register
2. From B's EUP, select the Routing tab, add an advance route:
When a call is received, My presence is in THESE STATES or... -> choose a state: Active on the phone
Route: Ring these numbers: enter C's URI

Setup:
1. A, B, C register
2. All users have MOH

Steps:
1. A call B
2. B answer call. Verify speech path
3. C call A.

Setup:
1. Configure a Hunt Group(linear) with 3 members.
2. Each of the 3 members register on Phones and are in the idle state.
3. At the SMB Interfase, set the No Answer Action = Continue to Hunt (from SMB Interface, select Provision > call answer groups, select the Hunt group, click on the ‘Show Advanced’ button, change the ‘No Answer Action’ to ‘Continue to Hunt’, press save Group button).

Steps:
1. Make a call from the PSTN to the Hunt Group.
2. Do not answer the call.
3. Terminate the call.

Steps:
1. Assign the Meetme (Office Suite Add-On) to a subscriber with Multimedia Office Premium
2. Subscriber A dials into meetme and login as chairperson and start the conference.
3. Subscriber B dials into the meetme as participant and enters the passcode

Steps:
1. A has voice mail service assigned from KBS portal
2. A, B register
3. B call A. A does answer. Verify that after timeout ringing at A, call is routed to A's voicemail.
4. B leave a voice message for A. Verify that A has a new voice mail notification
5. From A's phone, accessing to message box to connect to voicemail.