Iwr 0200 Ed01 - Tellabs Ont - Omnipcxenterpriser11.2 PDF

ALE Application Partner Program
Inter-Working Report
Partner: Tellabs
Application type: GPON-ONT
Application name: Tellabs ONT 7xx
Alcatel-Lucent Enterprise Platform:
OmniPCX Enterprise™
The product and release listed have been tested with the Alcatel-Lucent Enterprise Communication Platform and the
release specified hereinafter. The tests concern only the inter-working between the AAPP member’s product and the
Alcatel-Lucent Enterprise Communication Platform. The inter-working report is valid until the AAPP member’s product
issues a new major release of such product (incorporating new features or functionality), or until ALE International issues a
new major release of such Alcatel-Lucent Enterprise product (incorporating new features or functionalities), whichever first
occurs.
ALE INTERNATIONAL MAKES NO REPRESENTATIONS, WARRANTIES OR CONDITIONS WITH RESPECT TO THE

APPLICATION PARTNER PRODUCT. WITHOUT LIMITING THE GENERALITY OF THE FOREGOING, ALE
INTERNATIONAL HEREBY EXPRESSLY DISCLAIMS ANY AND ALL REPRESENTATIONS, WARRANTIES OR
CONDITIONS OF ANY NATURE WHATSOEVER AS TO THE AAPP MEMBER’S PRODUCT INCLUDING WITHOUT
LIMITATION THE IMPLIED WARRANTIES OF MERCHANTABILITY, NON INFRINGEMENT OR FITNESS FOR A
PARTICULAR PURPOSE AND ALE INTERNATIONAL FURTHER SHALL HAVE NO LIABILITY TO AAPP MEMBER OR
ANY OTHER PARTY ARISING FROM OR RELATED IN ANY MANNER TO THIS CERTIFICATE.
ALE Application Partner Program – Inter-working report - Edition 1 - page 1/153

Copyright © ALE International 2016
Certification overview
Date of the certification March 2016
ALE International representative Frank Gadot

AAPP member representative Les Murray
Alcatel-Lucent Enterprise OmniPCX Enterprise

Communication Platform
Alcatel-Lucent Enterprise
OXE R11.2 – L2.300.20e
Communication Platform release
705GR
729GP
AAPP member application release 702
704
All are version FP25.7.1
Application Category Gateway
Author(s): F. Gadot
Reviewer(s): R. Himmi
Revision History
Edition 1: creation of the document – March 2016
Test results
Passed Refused Postponed
Passed with restrictions
Refer to the section 0 for a summary of the test results.
IWR validity extension

AAPP Member Contact Information
Contact name TELLABS TAC SUPPORT

Title 24x7 Technical Support
Address 1415 WEST DIEHL ROAD

City NAPERVILLE, IL
ZIP 06825
Country USA
Phone +1.800.443.5555
WEB http://www.tellabs.com

TABLE OF CONTENTS
1 INTRODUCTION ...................................................................................................................................... 7
2 VALIDITY OF THE INTERWORKING REPORT ............................................................................. 8
3 LIMITS OF THE TECHNICAL SUPPORT ......................................................................................... 9
3.1 CASE OF ADDITIONAL THIRD PARTY APPLICATIONS ............................................................................. 9
4 APPLICATION INFORMATION ........................................................................................................ 10
5 TEST ENVIRONMENT .......................................................................................................................... 11
5.1 GENERAL ARCHITECTURE .................................................................................................................. 11
5.2 HARDWARE CONFIGURATION ............................................................................................................ 13
5.3 SOFTWARE CONFIGURATION .............................................................................................................. 13
6 SUMMARY OF TEST RESULTS ........................................................................................................ 14
6.1 SUMMARY OF MAIN FUNCTIONS SUPPORTED ...................................................................................... 14
6.2 SUMMARY OF PROBLEMS ................................................................................................................... 15
6.3 SUMMARY OF LIMITATIONS ............................................................................................................... 15
6.4 NOTES, REMARKS .............................................................................................................................. 15
7 TEST RESULT TEMPLATE ................................................................................................................ 16
8 TEST RESULTS – 705G ........................................................................................................................ 17
8.1 COMMISSIONING ................................................................................................................................ 17
8.2 AUDIO CODECS NEGOTIATIONS/ VAD / FRAMING .............................................................................. 19
8.3 OUTGOING/INCOMING CALLS ............................................................................................................ 21
8.4 FEATURES DURING CONVERSATION ................................................................................................... 27
8.5 CALL TRANSFER ................................................................................................................................ 29
8.6 VOICE MAIL ...................................................................................................................................... 32
8.7 HOTEL / HOSPITAL TESTS ................................................................................................................. 34
8.7.1 Provisioning .............................................................................................................................. 34
8.7.2 Check-in and check-out............................................................................................................. 35
8.7.3 Voice mail ................................................................................................................................. 36
8.7.4 Wake up..................................................................................................................................... 37
8.7.5 Change room state .................................................................................................................... 39
8.7.6 Mini-Bar.................................................................................................................................... 39
8.7.7 Multi occupation ....................................................................................................................... 40
8.7.8 Additional Tests. ....................................................................................................................... 41
8.8 ATTENDANT ...................................................................................................................................... 42
8.9 FAX .................................................................................................................................................. 44
8.9.1 Basic T38 Fax calls ................................................................................................................... 44
8.9.2 Codec & Multi algorithm feature .............................................................................................. 44
8.9.3 Passthrough mode between “SIP Devices” .............................................................................. 45
8.10 MODEM VBD .................................................................................................................................... 45
8.11 DUPLICATION AND ROBUSTNESS ....................................................................................................... 46
9 TEST RESULTS – 729 GP..................................................................................................................... 47
9.1 COMMISSIONING ................................................................................................................................ 47
9.5 CALL TRANSFER ................................................................................................................................ 59
9.6 VOICE MAIL ...................................................................................................................................... 62
9.7 HOTEL / HOSPITAL TESTS ................................................................................................................. 64
9.7.1 Provisioning .............................................................................................................................. 64

9.7.3 Voice mail ................................................................................................................................. 66
9.7.4 Wake up..................................................................................................................................... 67
9.7.6 Mini-Bar.................................................................................................................................... 69
9.7.7 Multi occupation ....................................................................................................................... 70
9.7.8 Additional Tests. ....................................................................................................................... 71
9.8 ATTENDANT ...................................................................................................................................... 72
9.9 FAX .................................................................................................................................................. 74
9.9.1 Basic T38 Fax calls ................................................................................................................... 74
9.9.2 Codec & Multi algorithm feature .............................................................................................. 74
9.10 MODEM VBD .................................................................................................................................... 75
9.11 DUPLICATION AND ROBUSTNESS ....................................................................................................... 76
10 TEST RESULTS - 702 ........................................................................................................................ 77
10.1 COMMISSIONING ................................................................................................................................ 77
10.5 CALL TRANSFER ................................................................................................................................ 89
10.6 VOICE MAIL ...................................................................................................................................... 92
10.7 HOTEL / HOSPITAL TESTS ............................................................................................................. 94
10.7.1 Provisioning .............................................................................................................................. 94
10.7.3 Voice mail ................................................................................................................................. 96
10.7.4 Wake up..................................................................................................................................... 97
10.7.6 Mini-Bar.................................................................................................................................... 99
10.7.7 Multi occupation ..................................................................................................................... 100
10.7.8 Additional Tests. ..................................................................................................................... 101
10.8 ATTENDANT .................................................................................................................................... 102
10.9 FAX ................................................................................................................................................ 104
10.9.1 Basic T38 Fax calls ................................................................................................................. 104
10.9.2 Codec & Multi algorithm feature ............................................................................................ 104
10.9.3 Passthrough mode between “SIP Devices” ............................................................................ 105
10.10 MODEM VBD .............................................................................................................................. 105
10.10.1 Passthrough mode between “SIP Devices” ........................................................................ 105
10.11 DUPLICATION AND ROBUSTNESS ................................................................................................. 106
11 TEST RESULTS - 704 ...................................................................................................................... 107
11.1 COMMISSIONING .............................................................................................................................. 107
11.2 AUDIO CODECS NEGOTIATIONS/ VAD / FRAMING ............................................................................ 109
11.3 OUTGOING/INCOMING CALLS .......................................................................................................... 111
11.4 FEATURES DURING CONVERSATION ................................................................................................. 116
11.5 CALL TRANSFER .............................................................................................................................. 119
11.6 VOICE MAIL .................................................................................................................................... 122
11.7 HOTEL / HOSPITAL TESTS ........................................................................................................... 124
11.7.1 Provisioning ............................................................................................................................ 124
11.7.2 Check-in and check-out........................................................................................................... 125
11.7.3 Voice mail ............................................................................................................................... 126
11.7.4 Wake up................................................................................................................................... 127
11.7.5 Change room state .................................................................................................................. 129
11.7.6 Mini-Bar.................................................................................................................................. 129
11.7.7 Multi occupation ..................................................................................................................... 130
11.7.8 Additional Tests. ..................................................................................................................... 131
11.8 ATTENDANT .................................................................................................................................... 132
11.9 FAX ................................................................................................................................................ 134
11.9.1 Basic T38 Fax calls ................................................................................................................. 134
11.9.2 Codec & Multi algorithm feature ............................................................................................ 134

11.9.3 Passthrough mode between “SIP Devices” ............................................................................ 135
11.10 MODEM VBD .............................................................................................................................. 135
11.10.1 Passthrough mode between “SIP Devices” ........................................................................ 135
11.11 DUPLICATION AND ROBUSTNESS ................................................................................................. 136
12 APPENDIX A: ONT CONFIGURATION REQUIREMENTS .................................................... 137
13 APPENDIX B: OMNIPCX ENTERPRISE CONFIGURATION REQUIREMENTS ............... 145
13.1 GLOBAL PARAMETERS..................................................................................................................... 145
13.2 CONFIGURATION OF THE SIP USERS................................................................................................. 145
13.3 PREFIX PLAN ................................................................................................................................... 145
13.4 SUFFIX PLAN ................................................................................................................................... 146
13.5 TONES PLAYED BY OXE .................................................................................................................. 146
14 APPENDIX C: AAPP MEMBER’S ESCALATION PROCESS .................................................. 147
15 APPENDIX D: AAPP PROGRAM ................................................................................................. 148
15.1 ALCATEL-LUCENT APPLICATION PARTNER PROGRAM (AAPP)....................................................... 148
15.2 ENTERPRISE.ALCATEL-LUCENT.COM .............................................................................................. 149
16 APPENDIX E: AAPP ESCALATION PROCESS ......................................................................... 150
16.1 INTRODUCTION ................................................................................................................................ 150
16.2 ESCALATION IN CASE OF A VALID INTER-WORKING REPORT ........................................................... 151
16.3 ESCALATION IN ALL OTHER CASES ................................................................................................... 152
16.4 TECHNICAL SUPPORT ACCESS .......................................................................................................... 153

1 Introduction
This document is the result of the certification tests performed between the AAPP member’s
application and Alcatel-Lucent Enterprise’s platform.
It certifies proper inter-working with the AAPP member’s application.
Information contained in this document is believed to be accurate and reliable at the time of printing.
However, due to ongoing product improvements and revisions, ALE International cannot guarantee
accuracy of printed material after the date of certification nor can it accept responsibility for errors or
omissions. Updates to this document can be viewed on:
- the Technical Support page of the Enterprise Business Portal

(https://businessportal.alcatel-lucent.com) in the Application Partner Interworking Reports
corner (restricted to Business Partners)
- the Application Partner portal (https://applicationpartner.alcatel-lucent.com) with free

access.

2 Validity of the InterWorking Report
This InterWorking report specifies the products and releases which have been certified.
This inter-working report is valid unless specified until the AAPP member issues a new major
release of such product (incorporating new features or functionalities), or until ALE International
issues a new major release of such Alcatel-Lucent Enterprise product (incorporating new features
or functionalities), whichever first occurs.
A new release is identified as following:

 a “Major Release” is any x. enumerated release. Example Product 1.0 is a major product
release.
 a “Minor Release” is any x.y enumerated release. Example Product 1.1 is a minor product
release
The validity of the InterWorking report can be extended to upper major releases, if for example the
interface didn’t evolve, or to other products of the same family range. Please refer to the “IWR
validity extension” chapter at the beginning of the report.
Note: The InterWorking report becomes automatically obsolete when the mentioned product
releases are end of life.

3 Limits of the Technical support
For certified AAPP applications, Technical support will be provided within the scope of the features
which have been certified in the InterWorking report. The scope is defined by the InterWorking
report via the tests cases which have been performed, the conditions and the perimeter of the
testing and identified limitations. All those details are documented in the IWR. The Business Partner
must verify an InterWorking Report (see above “Validity of the InterWorking Report) is valid and that
the deployment follows all recommendations and prerequisites described in the InterWorking
Report.
The certification does not verify the functional achievement of the AAPP member’s application as
well as it does not cover load capacity checks, race conditions and generally speaking any real
customer's site conditions.
Any possible issue will require first to be addressed and analyzed by the AAPP member before
being escalated to ALE International. Access to technical support by the Business Partner requires
a valid ALE maintenance contract
rd
For details on all cases (3 party application certified or not, request outside the scope of this IWR,
etc.), please refer to Appendix F “AAPP Escalation Process”.
3.1 Case of additional Third party applications

In case at a customer site an additional third party application NOT provided by ALE International is
included in the solution between the certified Alcatel-Lucent Enterprise and AAPP member products
such as a Session Border Controller or a firewall for example, ALE International will consider that
situation as to that where no IWR exists. ALE International will handle this situation accordingly (for
more details, please refer to Appendix F “AAPP Escalation Process”).

4 Application information
Application commercial name: GPON ONT 705GR
Application version: version: FP25.7.1
Interface type:
SIP Loosely Coupled : used to manage the supplementary services requiring Hook-Flash user
interaction (e.g. Call Waiting, Call Hold,etc). In this mode, hook-flash is handled locally at
the ONT level and not reported to the OmniPXC Enterprise.
Most switch feature activation/deactivation is performed by using Vertical Service Codes or

Feature Access Code (e.g *43# to activate Call Waiting service). These codes are handled
via standard digitmap provisioning at the ONT without usage of hook flash key. Call
Transfer, Call Waiting and Three Way Calling are all handled by the ONT and activated via
the Flash Key.
Brief application description:
The ONT includes the SIP User Agent that interfaces with the OmniPCX server via the SIP protocol.
This requires SIP interoperability for the different supported supplementary services that need specific
handling at the ONT level
In addition, it includes the Analog Terminal Adapter (ATA) that drives the physical line interface with
the subscriber telephone set (e.g Tones, Ringing, CLIP, etc). This interface requires a specific
customization in line with the country specifications and customer requirements.
Each ONT supports 2FX POTS ports which are integrated into the ONT.

5 Test environment
5.1 General architecture

See picture below for standard deployment application. For lab environment, a VPN remote
connection will be established between OLT and OmniPCX Enterprise.
Optical
Network
Terminals
(ONT)
1G or 1490nm
10G
Netwo Optical
rk Optical
Uplink
1310nm
Networ
s Optical k Line
Line Termina Termina
ls
Terminal l (OLT)
(ONT)
(OLT)

Tellabs 705G ONT
 Data, Voice, IP
Video, RF
Video
1490nm  (4) Gigabit
Ethernet
 (2) POTS
 Tellabs
(1) RF702
Video
ONT––
downstream
Outdoor Unit
 Data, Voice, IP
& RF Video
1310nm  (1) Gigabit
Ethernet
 (2) POTS
 (1) RF Video –
downstream

5.2 Hardware configuration
o OmniPCX Enterprise :
o Common Hardware
o Voice Mail 4645
o Single CPU call server
o Several ONTs configured with different combinations:
o Combination 1:
 One port configured as “SIP Extension” (SEPLOS) for analog phone
 One port configured as “SIP Device” for Fax machine
o Combination 2:
 both ports configured as “SIP Extension” (SEPLOS) for analog phones
o Analog Phones, Fax machines connected to the ONT
o Set up details:
o 1199: Voice mail 4645
o 9: prefix for external ISDN T1 calls via loopback
o Global compression type : G.711
o Law: U-law
o VoIP framing : 20ms
5.3 Software configuration

Alcatel-Lucent OXE OmniPCX Enterprise
o OXE R11.2-l2.300.20.e
Application platform:
 ONT705GR,ONT702, ONT704G, ONT729GP – all are running load FP27.0.1

6 Summary of test results
6.1 Summary of main functions supported

The main supported functions implied in the test campaign are:
705GR 702 729GP 704 Comment

Registration OK OK OK OK
Basic Call OK OK OK OK
Call
Forwarding
OK OK OK OK
Uncondition
al
Call Fwd requires special switch configuration.
Call
Forwarding OK OK OK OK
Busy
Call
Forwarding OK OK OK OK
No Answer
Caller ID
OK OK OK OK
(CLIP)
Caller Name Couldn’t test European special characters as couldn’t
OK OK OK OK enter them. ONTs just pass what is sent, no ONT
issues.
CLIR
OK OK OK OK
permanent
CLIR per call OK OK OK OK
Call Waiting OK OK OK OK .
Call Hold for
OK OK OK OK
Inquiry
3 way Call North American methods supported by all.
OK OK OK OK
Recall 1/2/3, 702 and 729 only.
Call Transfer North American methods supported by all.
OK OK OK OK
Attended Recall 1/2/3, 702 and 729 only.
Call Transfer North American methods supported by all.
OK OK OK OK
Unattended Recall 1/2/3, 702 and 729 only.
Distinctive Supported by all ONTs, but via standard model
Ringing (distinctive ring specified by standard strings Bellcore-
NOK NOK NOK NOK dr1 thru Bellcore-dr6). Not supported by switch, so not
possible. If switch added support for configurable
strings, would work fine.
Direct
Connect (Hot OK OK OK OK
Line)
Codec g.711 is the only supported codec.
(PCMU, 711 711 711 711
PCMA, G729
FAX (G711) OK OK OK OK
FAX (T38) OK OK OK OK
Modem Call NOK NOK NOK NOK

Direct
Connect OK NOK OK OK
(Warm Line)
RFC2833 OK OK OK OK
Voice mail OK OK OK OK 704 needs unsolicited notifies to work correctly.
Message Unsolicited Notify needed for 704 to work.
Waiting OK OK OK NOK All others support subscription.
Indicator
6.2 Summary of problems

None
6.3 Summary of limitations

 The configuration in OmniPCX Enterprise is specific for Analog phones and for
Fax Machines respectively “SIP Extension” and “SIP Device”.
 For this certification only one OXE CPU was used.
 G.711 is the only codec supported.
 OXE doesn’t support unsolicited Notify so no support for Message Waiting Indication (MWI)
on 704 ONLY. All others work.
 Fax transmission: T38 mode is supported, Passthrough mode is only supported between SIP
devices.
 Modem VBD is not supported neither on “SIP Extension” nor on “SIP Device” ports towards
PSTN. Pass-through mode is only supported between SIP devices.
 TLS is not supported by Alcatel-Lucent OXE R11.2 for SIP communication
 Tellabs ONTs 702, 704 and 705 in this test report only accept UDP communications. Tellabs
ONT 729 supports UDP, TCP and TLS.
 ONT 704 cannot support MWI due to the switch not supporting unsolicited NOTIFY
messages.
 ONT 704 cannot support g.711 aLaw.
 704 does not support DNS.
6.4 Notes, remarks

“SIP Extension” configuration is mandatory for services like Call Forwarding.
Call Waiting has to be enabled or disabled on the ONT which allows config via a profile.
Sip Device is required for configuration of T.38 fax lines.
The tests were done in a NON REDUNDANCY CONFIGURATION (only 1 CPU on OXE)

7 Test Result Template
The results are presented as indicated in the example below:
Test
Case Test Case N/A OK NOK Comment
Id
Test case 1
1  Action
 Expected result
Test case 2 The application

 Action waits for PBX timer
2
 Expected result or phone set hangs
up
Test case 3
Relevant only if the
 Action
3 CTI interface is a
 Expected result
direct CSTA link
Test case 4
 Action No indication, no
4
 Expected result error message
… …
Test Case Id: a feature testing may comprise multiple steps depending on its complexity. Each step
has to be completed successfully in order to conform to the test.
Test Case: describes the test case with the detail of the main steps to be executed the and the
expected result
N/A: when checked, means the test case is not applicable in the scope of the application
OK: when checked, means the test case performs as expected
NOK: when checked, means the test case has failed. In that case, describe in the field “Comment”
the reason for the failure and the reference number of the issue either on Alcatel-Lucent side or on
Application Partner side
Comment: to be filled in with any relevant comment. Mandatory in case a test has failed especially
the reference number of the issue.

8 Test Results – 705G
8.1 Commissioning
These tests shall verify that the analog phones connected to the ONT can be declared on OmniPCX
Enterprise as SIP extension (SEPLOS) and are able to register properly on the OXE server registrar.
Test
Id
SIP set registration using static IP
Address
Configure all your SIP sets from
MCDU number SIPset-1 to SIPset-8
to register with the OXE IP address
<Node1_Main_IP@>. All Tellabs ONTs only support
1 DHCP for IP address
Check the registration on your sets assignment.
and the display
Note that authentication is disabled

for these users, the password doesn’t
matter.
SIP set registration, using DNS,
without authentication
The phone SIPset-1 is configured with
Primary DNS IP address:
<Primary_DNS_IP> and Secondary
DNS IP address
2
<Secondary_DNS_IP>.
And SIP proxy is declared with DNS
name : <DNS_Name>.
Check the phone registration and

display.
DHCP registration
3
(with OXE internal DHCP server)
NTP registration
The SIP phone SIPset-1 is configured
to retrieve the date and time from
OLT performs NTP, time sent
the OXE Node 1 main IP address
to ONT via ME, verified correct
4 <Node1_Main_IP@>.
on caller ID.
705 Pass
Check the phone retrieves the right
date and time information and
displays it.
Support of “423 Interval Too
Brief” (1)
The phone SIPset-1 is configured with By default,
5 a value lower than 1800 seconds. OXE>SIP>SIP Registrar>Min
Expires=1800s
display.

SIP set registration with
authentication
Enable SIP authentication on the OXE
system.
Configure the phone SIPset-1 with

authentication password 1234.
6
display.
Redo the same tests with a wrong

password and check that the phone is
rejected.
Signalling TCP-UDP.
If applicable configure your SIP set to
use the protocol Sip over UDP and SIP
7 over TCP
In the two cases, check the

registration, and basic calls.
Notes:
(1) On the SIP client, specify a default registration period inferior to that of OXE SIP registrar. OXE will
reject with error “423 Interval Too Brief”. Check that SIP set increases registration period accordingly.

8.2 Audio codecs negotiations/ VAD / Framing
These tests check that the phones are using the configured audio parameters (codec, VAD,
framing).
Base Configuration :
Phones:
Configure the phones to use G.711 A-law, G.711 mu-law, G.729 in this order.
Configure the phones to use framing=20ms (G.711 and G.729)
Configure the phones to NOT use VAD.
OXE :
Manage 2 IP domains:
Domain1: intra=no compression extra=compression
Assign SIPset-1 and SIPset-2 to domain 1.
Assign SIPset-3 to domain 2.
Sip or rtp
Set system compression type = G.729
Note: ONT and OXE will use 20ms framing.
Test
Id
Call from SIPset-1 to SIPset-2 (intra-
domain)
Check that the call is established
1
using direct RTP in G711 A-law.
Check audio quality
Call from SIPset-1 to SIPset-3

g.729 is not supported.
(extra-domain)
Low bw codecs not
2 needed, system is GigE at
using direct RTP in G729.
all points.
Check audio quality
Set system law = mu-law

Configure the phone to use G.711
mu-law, G.711 A-law, G.729 in this
order
Consistently score above a
3 Call from SIPset-1 to SIPset-2 (intra-
MOS of 4.
domain)
using direct RTP in G711 mu-law.
Check audio quality
Configure SIPset-1 to use VAD


4 domain) VAD not supported.
Check audio quality

Configure SIPset-1 NOT to use VAD
Redo the same tests


(extra-domain)
using direct RTP in G729. G729 not supported
5
Check audio quality

Redo the same tests

8.3 Outgoing/Incoming Calls
The calls are generated to several users belonging to the same OXE.
Called party can be in different states: free, busy, out of service, etc.
Calls to data devices are refused.
Points to be checked: tones, voice during the conversation, display (on caller and called party), hang-
up phase.
SIP authentication will be activated on OXE and used by default for all further testing.
Test
Id
Local call
With SIPset-1, call SIPset-2.
Check that SIPset-2is ringing and

1 answer the call.
Check the display and audio during all

steps (dialing, ring back tone,
conversation, and release).
Local call with overlap dialing
but,
Dial a first part of the number: 5xx,
2
pick up, wait one second and dial 09.
Check that call is transmitted to

SIPset-2.
External call to SIP terminal (via

T2 Loopback)
Check that external call back number is
shown correctly
With SIPset-2 dial 9xxxx (prefix to take
T2 loop followed by the 4 last digits of
3 SIPset-1)
Check that SIPset-1 is ringing and the

external call number is shown correctly
Take the call and check audio, display

calling name and call release.
Display: Call to free SIP terminal
from user with a name containing
non-ASCII characters. Check caller
display. Switch won’t accept the
characters at Telnet interface,
With SIPset-2 call SIPset-1 (extension
can’t test. The line will send
4 with a name containing non-ASCII
whatever is in the Display name.
characters).
Will work on all ONTs.
Check that SIPset-1 is ringing and
check on its display the name
SIP_SIPset-2_éëêèè is printed.

Test
Id
Check that non-ASCII characters

(éëêèè) are correctly printed.
from user with a UTF-8 name
containing non-ASCII characters.
Check caller display.
with a name containing UTF-8
characters).
5
check on its display the name:
SIP_SIPset-7_&@@@### is printed.
Check that UTF-8 characters

(@@@###) are correctly printed.
CLIR permanently provisioned (by

OXE configuration)
Activate the service via Entity -> Caller
ID secret: True
With SIPset-2 call SIPset-1.
Check that SIPset-1 is ringing, take the

call and check that SIPset-2 identity is
hidden.
Do the same test but by calling an

6 external PSTN number.
Deactivate the service via Entity ->

Caller ID secret: False

NOT hidden.

external PSTN number.
CLIR controlled per call
With SIPset-2 call SIPset-1 by dialing

*31*SIPset-1# (dial *31* + <target
7 MCDU number>#) in order to hide
SIPset-2 identity.
Check that SIPset-1 is ringing, take

the call and check that SIPset-2
identity is hidden.

Test
Id
Call to local user with no answer.
With SIPset-1, call SIPset-2
And don’t answer the call.

Check time out and display.
8 After the call being released, place a

new call to SIPset-2, answer the call
and check that audio, display name is
ok.
Note that SIPset-2 don’t have a Voice

Mail
Call waiting
With SIPset-1 call SIPset-2 and answer

the call.
With SIPset-3 call SIPset-2 and check
that the ringing tone is received by
SIPset-3.
Check that the Call Waiting tone is

served to SIPset-2.
9
On SIPset-2, release active call with
“R”+SOC 1
Verify that SIPset-2 is connected to
the waiting call
When a call is waiting, dial “R”+SOC 2

instead of “R” + SOC 1.
Verify that the active call is put on hold
and switch to the waiting call.
Call to busy user

(SIP: “486 Busy Here”)
the call.
With SIPset-3 call SIPset-2 and wait to
make it busy
10
With SIPset-4 call SIPset-2 which is busy
Check the busy tone and display.
Do the same test with a PSTN incoming

call.
Call to user in “Do not Disturb”
state
(SIP: “603 Decline”)
11
Dial “42” on the SIPset-1 in order to
enable the DND. Wait for
acknowledgement ring back tone from

Test
Id
OXE
With the SIPset-2 call the SIPset-1.
Check ring back tone and display.
Redial 42 on SIPset-1 to cancel the

DND
Distinctive Dial tone when Call
Forwarding activated
On SIPset-1, dial the *21*SIPset-2#
(immediate forwarding prefix +
<target MCDU number> + #) to
activate the CFU. Wait for
12
OXE.
Off Hook SIPset-1, and verify the tone
Call to local user, immediate

forward (CFU).
(SIP: “302 Moved Temporarily”)
OXE.
13
With SIPset-3 call the SIPset-1.
display. Take the call check audio and
hung up.

call.
Dial #21# on SIPset-1 for forward

cancellation.
Call to local user, forward on no
reply (CFNR).
(Forward on no reply prefix +<target
MCDU number> + #) to activate the
CFNR. Wait for acknowledgement ring
back tone from OXE.
14 With SIPset-3 call the SIPset-1. Check

that SIPset-1 is ringing but don’t take
the call and wait the time out (about 30
sec).
After time out, check that SIPset-2 is

ringing and take the call.
Check the audio and display.

Test
Id

call.

cancellation.
Call to local user, forward on busy
(CFB).
(forward on busy prefix +<target
CFB. Wait for acknowledgement ring
back tone from OXE.
With SIPset-3, call SIPset-1 and take the

call.
With SIPset-4, call SIPset-1 and wait to Camp On has to be manually
15 make SIPset-1 busy disabled on Dynamic State for
With SIPset-5, call SIPset-1. this to work.
Check that SIPset-2 is ringing and take

the call.

call.

cancellation.
Call Back on free set
From SIPset-1 call SIPset-2
Dial “5” (Call Back suffix) while SIPset-
16 2 is ringing and release the call. Not applicable to analog phone.
Activate the call back from SIPset-2.
Check that SIPset-1 is ringing, answer
the call and check audio + display.
Call Pick Up.
While SIPset-2 is ringing, pick-up the
call from SIPset-3 by dialing the pick-
17
up prefix + SIPset-2
Check audio and display.
Distinctive ringing
Call to SIPset-1 from SIPset-2, check
that internal ringing is applied to
SIPset-1
Alcatel-Lucent OXE does not
18
support this feature for SIP
Place an ISDN incoming call to SIPset-
1,
check that external ringing is applied to
SIPset-1
Direct Connect Hot Line
19
On-Hook from SIPset-1

Test
Id
Check that automatic call is established
to user configured in the ONT
Direct Connect Warm Line

Check that SIPset-1 can dial some
20 number or the automatic call to the
configured number takes place after a
predefined timer expires.
SIP session timer expiration
Check if call is maintained or released
after the session timer has expired:
With SIPset-1 dial SIPset-2.
21 Take the call on SIPset-2 and wait for

time out expiration (about 30 minutes).
Check that call is maintained or

release.
Local call to unplugged SIP terminal

Unplug the SIPset-2
From SIPset-1, call SIPset-2 while If line is registered but

SIPset-2 still registered unresponsive forward to voice
mail after 3 failed invites.
Subsequent calls go
Check the ring back tone and display
22 immediately to voice mail. This
case works.
From SIPset-1, call SIPset-2 once SIPset-
If I unregister as shown in the
2 not registered anymore nd
2 part of the use case lines
SIP: “480 Temporarily Unavailable” will will give reorder/fast busy.
be sent
Check out of service ? tone

Call to wrong number
(SIP: “404 Not Found”)
With the SIPset-1 call a wrong number
23
which is not in the dialing plan.
Check the ring back tone and display.

Call rejected by call handling
(SIP: “183 Progress/487 Request
24
Terminated”)
e.g. max number of calls reached etc.

8.4 Features during Conversation
Features during conversation between local user and SIP user must be checked.
Check that right tones are generated on the SIP phone.
Test
Id
Enquiry call using “R” key
With SIPset-1 call SIPset-2
Take the call, check audio and display.
From SIPset-1 put SIPset-2 on hold

with “R” key, check tones and display
on both sets, then press again “R” key
+ 1 to resume the call

Only North American style via
1 with “R” key.
flashook is supported on the
Check dial tone and dial SIPset-3 705GR.
Answer the call and check audio and
display.
After enquiry call, dial “R”+1 to

release the current call and check
that SIPset-1 and SIPset-2 are in
conversation.
Broker call using “R”+2 key

After enquiry call, dial “R”+2 several
Only North American style via
times to toggle between the two
2 flashook is supported on the
calls
705GR.
Call park:
With SIPset-1 call SIPset-2 and take the
call.
On the SIP set SIPset-1 park the call
with SIPset-2 by dialing the 402 SIPset-
3 (402 call park prefix + <target
MCDU number>)
Check that OXEset-2 is put on hold. Only North American style via
3 flashook is supported on the
Release SIPset-1 but not SIPset-2. 705GR.
Check that SIPset-2 is still on hold.
From SIPset-4 dial 402SIPSet-3 (402

call park prefix + <SIPSet-3
number>).
Check that SIPset-4 and SIPSet-2 are
in conversation + display.
Send/receive DTMF
Configure SIPset-1 to send DTMF
using RFC 2833
4
From SIPset-1 call the Voice Mail at
14999 and try to navigate in its menu

Test
Id
listed by the voice guide.
Check that you can navigate in the
menus
Three party conference
After enquiry call, dial “R” + 3 key to

activate three party conference.
Press Hookflash and then 1.The last

active call is released and the other
call is activated.
5
Do the test with all 3 participants
being SIPsets.
Do the test with 1 participants being

an external PSTN user.


8.5 Call Transfer
During the consultation call step, the transfer service can be requested and must be tested.
Several transfer services exist: Unattended Transfer, Semi-Attended Transfer and Attended Transfer.
Audio, tones and display must be checked.
We use the following scenario, terminology and notation:
There are three actors in a given transfer event:

 A – Transferee: the party being transferred to the Transfer Target.
 B – Transferor: the party doing the transfer.
 C – Transfer Target: the new party being introduced into a call with the Transferee.
There are three sorts of transfers in the SIP world:

 Unattended Transfer or Basic Transfer: The Transferor provides the Transfer Target's
contact to the Transferee. The Transferee attempts to establish a session using that contact
and reports the results of that attempt to the Transferor.
Note: Unattended Transfer is not applicable
 Semi-Attended Transfer or Early Attended Transfer or Transfer on ringing:

1. A (Transferee) calls B (Transferor).
2. B (Transferor) calls C (Transfer Target). A is on hold during this phase. C is in ringing
state (does not pick up the call).
3. B executes the transfer. B drops out of the communication. A is now in contact with C, in
ringing state. When C picks up the call it is in conversation with A.
Note: This is Call Transfer Unattended/Blind per access/ONT ? terminology.
 Attended Transfer or Consultative Transfer or Transfer in conversation:

2. B (Transferor) calls C (Transfer Target). A is on hold during this phase. C picks up the
call and goes in conversation with B.
3. B executes the transfer. B drops out of the communication. A is now in conversation with
C.
Detailed usage of hook flash (RR) to operate this service is provided in http://ct.web.alcatel-
lucent.com/scm-lib4/show-entry.cgi?number=3FC-40122-W301-TQZZA
(Section 4.5)

8.5.1.1 Semi-attended transfers
The scenario is the following:
B has Call Transfer service. User A calls user B. B answers. A and B are connected. During the
conversation B presses Hookflash to put A on hold and dials user C. B hears normal call progress
tones including ringback. A is on hold. B on-hook while ringing. A and C are connected.C releases
the call.
Test Test Case N/A OK NOK Comment

Case
Id
A B C
Transferee Transferor Transfer Target
Type MCDU Type MCDU Type MCDU

Number Number Number
1 Ext. Call N/A SIP SIPset-1 SIP SIPset-2
2 SIP SIPset-1 SIP SIPset-2 Ext. Call N/A
3 SIP SIPset-1 SIP SIPset-2 SIP SIPset-3

8.5.1.2 Attended Transfer (in Conversation)
conversation B presses Hookflash to put A on hold. B dials C. B hears normal call progress tones
including ringback. A is on hold. C picks up the phone and B and C communicate. B presses
Hookflash and then 4, hears tone and hangs up (the only allowed action in such state). A and C are
connected.

Case
Id
A B C


8.6 Voice Mail
Voice Mail notification, consultation and password modification must be checked.
MWI (Message Waiting Indication) has to be checked.
The Voice Mail number is 14999, and this service is enable on SIP sets SIPset-1, SIPset-2. The default
password is 000 for all accounts.
For these tests, DTMF sending (RFC 2833) has to be validated in order to use Voice Mail menu.
Note 1: explicit subscription is required for RFC3842 MWI (ref 2.8.24.7 SIP Endpoint developpers
guide)
Note 2: the directory number of the voice mail (14999) must be configured in the ONT
Test
Id
Message display activation, MWI
With SIPset-2 call the Voice Mail at
14999.
Follow the instructions in order to send
a voice message in SIPset-1 box.
1 Check that the MWI (Audible Tone) is

activated when SIPset-1 goes Off-
Hook.
Check that the Visual Message

waiting indicator is activated (LED
blinking)
Message consultation
14999.
Follow the instructions in order to listen
the voice message leaved during the
previous test. Check that you can listen
2
it and remove it.
Check that MWI (Audible Tone) and

Visual Message Waiting indication is
disabled on SIPset-1 after message
removal.
Password modification
14999 and follow the Voice guide in
order to modify the default password.
When modification is accepted

hangup.
3
Recall the voice mail and try to log
with a wrong password. Check the
rejection.

with the right password. Check the
service access.

Call to user forwarded to Voice
Mail
Forward the SIPset-2 to Voice Mail by
dialing 5114999 (51 prefix + <Voice
Mail number>).
4 With SIPset-1 call SIPset-2 and check

that you are immediately forwarded to
Voice Mail.
Check that you can leave a message
On SIPset-2 disable Voice Mail

forwarding with 41 prefix.
Voice mail deposit
Dial “6” (Voice Mail deposit suffix)
while SIPset-2 is ringing. Leave a
5
message when connected to the voice
mail and release the call.
Check the voice message on SIPset-2.

8.7 Hotel / Hospital tests
These tests check the phones behavior for SIP phone specific Hotel/Hospital features like
provisioning, check-in and out, do not disturb, wake-up, room state modification, mini bar, forward,
auto assignation and calls.
These tests check the phone provisioning as a room, suite, administrative or booth set.
8.7.1 Provisioning
Test
Id
Declare the SIP phone as a hotel
room set
In the OXE configuration, the SIP set

is declared as a hotel room set
Or
1
The SIP set is configured (user and
password) with the parameters of an
already declared SIP hotel room set
The phone registers correctly to the

OXE.
administrative set

is declared as a hotel administrative
set
Or
2
already declared SIP hotel
administrative set

OXE.
booth set

is declared as a hotel booth set
Or
3
already declared SIP hotel booth set

OXE.

8.7.2 Check-in and check-out
These tests check the phone behavior after a check-in and a check-out as normal or VIP guest.
Test
Id
A client checks in a room
containing the SIP phone
In the OXE hotel menu (hotmenu), a

check-in is done and the client gets
1 the SIP phone room
Or
already declared SIP hotel room set in
which a client has already checked-in.
A client checks in as VIP in a room
Same as above but with the VIP

2 parameter set.
When this phone calls a hotel
administrative set, the name
displayed is completed with specific
information.
A client checks out from a room
3
check-out is done for the client using
the SIP phone room

8.7.3 Voice mail
These tests check the phone behavior when interworking with the OXE 4645 voicemail.
In order to do these tests, the Room has to be checked in. If the room isn’t checked-in there is no
voicemail available. In the User parameters, make sure that before the Check-in there is no
Voicemail number.
Test
Id
Voice mail message signalization.
The voice mail (OXE 4645) number is

configured in the phone. Call The
1 phone and leave a message to its
voice mail (for example by forwarding
the phone to the voice mail). Check
that the message is indicated on the
phone (led or display).
Voice mail message listening.
The phone has a voice mail message

(see above). Press the voice mail key
2 and interacts with the voice mail to
listen to the message.
Check the led or display does not
show any new message as soon as
the last one is read.

8.7.4 Wake up
These tests check the phone behavior in case of "Wake up" activation / deactivation (on the phone
itself or from an administrative phone). To set a wake-up call it is necessary to dial the extension
mentioned in the test case and then after hearing the tone dial: “HH MM XXXX”. With “HH MM”
being the time of the alarm call and “XXXX” being the room extension number or the guest number.
After setting up the alarm call an audio message should confirm the time set. To cancel an alarm
call, dial the necessary extension mentioned in the test case and after hearing the tone dial the
extension aimed by the cancellation.
Test
Id
Wake Up is activated on the room
SIP phone
On the room SIP phone the Wake Up

1
is activated thanks to the prefix 506
When the wake up time arrives, the
phone rings. When the picked up, the
voice guide is played.
Wake Up is deactivated on the
room SIP phone

2
is deactivated thanks to the prefix 507
When the previous wake up time
arrives, nothing appends on the
phone set.
Suite wake Up is activated on a
suite SIP phone
On the suite master SIP phone the

suite wake Up is activated thanks to
the prefix 584
3
entire suite phones are ringing. When
picking up on one phone, the voice
guide is played and all the other
phones stop ringing.
Test also the activation from slave

suite phones. Behavior is the same.
Room wake Up is activated on a
suite SIP phone

the prefix 506
When the wake up time arrives, only
4
the suite phone on which the wake up
has been set is ringing. All the other
suite phones do not. When picking up,
the voice guide is played.

Suite wake Up is deactivated on a
5
suite SIP phone

Test
Id

Wake Up is deactivated thanks to the
prefix 585
phone sets (master and slaves).
Test also the deactivation from slave


8.7.5 Change room state
These tests check the phone can change the room state (as a room or administrative set).
Test
Id
The room state is changed on the
room SIP phone set.
On the room SIP phone the room

state is changed thanks to the prefix
587 (then made personal code, then
1 room status).
Check the new room state thanks to
the hotel menu (hotmenu).
Several room states are tried (1 =
done and available, 2 = to do
completely, 3 = to do partially, 4 to 9 =
problem).
8.7.6 Mini-Bar
These tests check the phone can change the "mini bar" state (as a room or administrative set).
Test
Id
The min-bar state is changed on
the room SIP phone set.
On the room SIP phone the mini-bar

588
1 Several mini-bar states are tried.
This test can be validated when

analyzing the hotel traces on the
OXE. The only verification done is to
see if the correct digits are sent to the
OXE.

8.7.7 Multi occupation
These tests check the phone behavior when several guests are located in the same room.
Test
Id
SIP set registration to OXE
1 The phone successfully registers to

the OXE
Incoming call to the room phone

number
2 Another phone (IPTouch) calls the

room phone number. The call can be
picked up and is successfully
established.
Incoming call to the first guest
phone number

first guest phone number. The call
can be picked up and is successfully
established.
Incoming call to the second guest
phone number

second guest phone number. The call
established.
Outgoing call by the first guest
In case of an external call (to a PSTN

5 user for example), the first guest
makes an outgoing call using his
guest ID. The call is successfully
established.
Outgoing call by the second guest

6 user for example), the second guest
established.

8.7.8 Additional Tests.
These tests check the good behavior in hotel suite mode.
Test
Id
For a suite with several phones,
check behavior when there is an
incoming call (to the suite phone
1
number) while there is already a
phone in conversation
Check that after a checkout all the

guest specific information are erased
(voice mail messages, phone state
2
like forward, do not disturb, wake up
time)
Test that DID attribution is working

3
fine with the SIP hotel sets

8.8 Attendant
An attendant console is defined on the system. Call going to and coming from the attendant console
are tested.
Test
Id
SIP set call to attendant
With SIPset-1, call attendant with prefix
9 (attendant call prefix), attendant
1 answers.
Check ringing back tone, display and

audio.
2nd incoming call to SIP set while
in conversation with attendant.
SIPset-1 being in conversation with the
attendant.
Make an ISDN incoming call to SIPset-

1 and try to answer the call.
2
The behaviour will depend on the
SIPset: either the second call can be
answered or not. Please precise the
behaviour of the set.
In any case, check that the call is

properly managed.
Attendant transfers SIP call , semi-
attended

answers.
3 From the attendant, call SIPset-2 and

transfer semi-attended.

display.
Redo the test by dialing from

attendant External ISDN call.
Attendant transfers SIP call,
attended

4 answers.
From the attendant, call SIPset-2 and

transfer attended.

display.

Attendant transfers External call ,
semi-attended
External ISDN call to attendant,

attendant answers.
5

display.
Attendant transfers External call,

attended

attendant answers.
6
transfer attended.

display.

8.9 FAX
Perform fax transmission in T.38 mode.
The port on which the Fax machine is connected must be declared as “SIP Device”.
Note: Passthrough mode is only supported between “SIP Devices” ports.
8.9.1 Basic T38 Fax calls
Test
Id
Send a fax from ONT Fax to another
ONT Fax
1
Redo the same test with a fax of 5
pages
Send a fax from ONT Fax to an
other ONT Fax via PSTN
loopback(corresponds to
2 sending/receiving Fax over PSTN)

pages
8.9.2 Codec & Multi algorithm feature
Test
Id
Configure the ONT Fax (G711)
Enable the Multi Algorithms for

compression OmniPCX Enterprise
1 system feature
Send a fax from ONT Fax to ONT Fax

via PSTN loopback
Configure the ONT Fax to allow

G729
2 G729 not supported
via PSTN loopback

G723
via PSTN loopback

8.9.3 Passthrough mode between “SIP Devices”
Test
Id
other ONT Fax in Passthrough mode
G.711 is Passthrough.
1
Same test as 6.9.2.
pages
8.10 Modem VBD

Modem VBD is not supported towards PSTN.
Test
Id
Not likely to work reliably in
Send a modem from an ONT to an most scenarios without a
other modem connected on an ONT timing input that is not
in Passthrough mode available in most locations.
1
For reliable modem
transport, the system needs
a composite clock input to
the OLT.

8.11 Duplication and Robustness
Note: only redundancy within the same IP subnetwork is in the scope. This should be transparent for
the ONT. Spatial redundancy is not supported by the ONT.
Check how the system will react in case of a CPU reboot, switchover or link failure etc.
Test
Id
Switchover to standby call server
1 Check that existing calls Not tested
local/external are maintained.
Not Tested
2 Check that an outgoing call can be
done just after the switchover.
3 Check that an existing call can be Not Tested
updated (put on hold, transfer…).
ONT reboot
Check that calls are possible as soon Remote reboot ONT and then
as ONT has come back to service. measure time to dial tone.
6
705 Pass (0:46s)
Do the test without or with active
conversations when ONT is rebooted.
Prior to registration timeout,
Temporary IP Link down with the
dial tone, then fast busy on
OXE
dial complete. After
Check that the ?? tone is presented
registration timeout
7 to the user
immediate fast busy.
Check that incoming/outgoing calls
Took several minutes after
are possible once the link is
recovery for the PBX to begin
reestablished.
forwarding invites to lines.

9 Test Results – 729 GP
9.1 Commissioning
Test
Id
Address
<Node1_Main_IP@>. All Tellabs ONTs only
1 support DHCP for IP
Check the registration on your sets address assignment.
and the display

matter.
DNS IP address
2
<Secondary_DNS_IP>.
name : <DNS_Name>.

display.
DHCP registration
3
NTP registration
OLT performs NTP, time
sent to ONT via ME, verified
4 <Node1_Main_IP@>.
correct on caller ID.
displays it.
Brief” (1)
The phone SIPset-1 is configured with By default,
5 a value lower than 1800 seconds. OXE>SIP>SIP Registrar>Min
Expires=1800s
display.

authentication
system.

6
display.

rejected.
Signalling TCP-UDP.
use the protocol Sip over UDP and SIP UDP, TCP, TLS supported
7 over TCP by 729GP. TLS Not
supported by OXE.
Notes:

framing).
Phones:
OXE :
Sip or rtp
Test
Id
domain)
1
Check audio quality

(extra-domain)
G729 not supported by
2 Tellabs
Check audio quality

order
Capture Provided, will include
system test reports of MOS
and PSQM. Consistently
domain)
score above a MOS of 4.
Check audio quality


Check audio quality

Redo the same tests


(extra-domain)
using direct RTP in G729. G729 not supported.
5
Check audio quality

Redo the same tests

up phase.
Test
Id
Local call

1 answer the call.

but,
2 Overlap dialing not applicable

SIPset-2.

T2 Loopback)
shown correctly
3 SIPset-1)


non-ASCII characters. Check caller
display.
4 with a name containing non-ASCII
characters).


Test
Id

With SIPset-7 call SIPset-1 (extension Note: Caller ID phones wont
with a name containing UTF-8 display some characters, show
characters). as blanks. Test set verifies
5 output matches display name
Check that SIPset-1 is ringing and and is properly formatted. (i.e.
check on its display the name: verified ONT is correct in all
SIP_SIPset-7_&@@@### is printed. cases, problem is phones)


OXE configuration)
ID secret: True

hidden.
ONT retrieves CLIR information
from “anonymous” in userinfo
of SIP From header and displays
“Unknown Number”.

NOT hidden.

With SIPset-2 call SIPset-1 by dialing

*31*SIPset-1# (dial *31* + <target
7 MCDU number>#) in order to hide
SIPset-2 identity.

identity is hidden.

Test
Id


ok.

Mail
Call waiting

the call.
SIPset-3.
Used N. American method.
Check that the Call Waiting tone is Recall button method only
served to SIPset-2. supported by 702 and 729.
9
Call Waiting is provided by the
“R”+SOC 1
ONT.
the waiting call

Call to busy user

the call.
ONT returns a 486 Busy Here
With SIPset-3 call SIPset-2 and wait to
message. Ignore 404 not found
make it busy
in trace, switch had lingering
10
registration for 1107 that hadn’t
With SIPset-4 call SIPset-2 which is busy timed out.

call.
Call to user in “Do not Disturb” Do not disturb results in voice
state prompt telling you to call back
(SIP: “603 Decline”) later(due to PBX config). Can
11
Dial “42” on the SIPset-1 in order to successfully enable and disable
enable the DND. Wait for DND feature works on all
acknowledgement ring back tone from ONTs.

Test
Id
OXE

DND
12
OXE.

forward (CFU).
acknowledgement ring back tone from On our switch Immediate
OXE. Forward is *60, clear is *64.
Switch side feature.
13
hung up.

call.

cancellation.
reply (CFNR).
back tone from OXE. *62 is CFNA
When on, forwards, when off,
goes to voicemail. *64 is
cancel.
sec).


Test
Id

call.

cancellation.
(CFB).
back tone from OXE.

call. Camp On has to be manually
With SIPset-4, call SIPset-1 and wait to disabled on Dynamic State for
15 make SIPset-1 busy this to work.

the call.

call.

cancellation.
Call Pick Up.
17
Distinctive ringing
SIPset-1
18
1,
SIPset-1
19

Test
Id

Clarified that Re-Invite is
SIP session timer expiration acceptable.
Check if call is maintained or released Re-invite only initiated by
after the session timer has expired: switch. Switch will terminate
when Re-invite not answered.
With SIPset-1 dial SIPset-2.
ONT will remain in active call
state until onhook. Can make
calls normally after onhook.
With Re-invite that is all you
can do.
Check that call is maintained or
release. 705 Pass

Unplug the SIPset-2
If line is registered but
From SIPset-1, call SIPset-2 while unresponsive forward to voice
SIPset-2 still registered mail after 3 failed invites.
Subsequent calls go
Check the ring back tone and display immediately to voice mail. This
22
case works.
From SIPset-1, call SIPset-2 once SIPset-
2 not registered anymore If I unregister as shown in the
nd
SIP: “480 Temporarily Unavailable” will 2 part of the use case lines
be sent will give reorder/fast busy.
729GP gives regular busy.
23

Call rejected by call handling
(SIP: “183 Progress/487 Request
24
Terminated”)
e.g. max number of calls reached etc.


Test
Id

North American behavior (flash
on Call Waiting) supported by
1 with “R” key.
all. Recall 1 Supported by 702
Check dial tone and dial SIPset-3 and 729..
display.

conversation.

2 hook) and Recall button
calls
behavior supported.
Call park:
call.
MCDU number>)
Check that OXEset-2 is put on hold. North American behavior (flash
3 hook) and Recall button
Release SIPset-1 but not SIPset-2. behavior supported.

number>).
Send/receive DTMF
using RFC 2833
4 From SIPset-1 call the Voice Mail at

menus
Three party conference For North America, it is A calls
5 B, flash B calls C flash, 3 way
After enquiry call, dial “R” + 3 key to conference. Digits after the

Test
Id
activate three party conference. flash do not control legs of
conference.
Check audio and display. North American behavior (flash
hook) and Recall button
Press Hookflash and then 1.The last behavior supported.
call is activated.

being SIPsets.



9.5 Call Transfer




C.
(Section 4.5)

the call.

Case
Id
A B C

Number Number Numbe
r

connected.

Case
Id
A B C



9.6 Voice Mail
guide)
Test
Id
14999.

Hook.

blinking)
14999.
2
it and remove it.

removal.

hangup.
3
rejection.

service access.

Mail
Mail number>).

Voice Mail.

Voice mail deposit
5

9.7.1 Provisioning
Test
Id
room set

Or
1

OXE.
administrative set

set
Or
2
administrative set

OXE.
booth set

is declared as a hotel booth set
Or
3

OXE.

Test
Id

Or

2 parameter set.
information.
3
the SIP phone room

9.7.3 Voice mail
Voicemail number.
Test
Id



9.7.4 Wake up
Test
Id
SIP phone

1
room SIP phone

2
phone set.
suite SIP phone

the prefix 584
3

suite SIP phone

the prefix 506
4

5
suite SIP phone

Test
Id

prefix 585


Test
Id
room SIP phone set.

1 room status).
problem).
9.7.6 Mini-Bar
Test
Id
On the room SIP phone the mini-bar Note: Due to description in

state is changed thanks to the prefix docs, just requires that we
588 pass VSC *3 then connect.
1 Several mini-bar states are tried. Connected but no voice
prompt. After that we just
This test can be validated when pass digits same as any call.
OXE.

Test
Id

the OXE

number

established.
phone number

established.
phone number

established.

established.

established.

Test
Id
1

(voice mail messages, phone state This is a switch function and
2
like forward, do not disturb, wake up doesn’t involve the endpoint.
time)
Test that DID attribution is working This is a switch function and

3
fine with the SIP hotel sets doesn’t involve the endpoint.

9.8 Attendant
are tested.
Test
Id
1 answers.

audio.
attendant.

2

properly managed.
attended

answers.


display.

attended

4 answers.

transfer attended.

display.

semi-attended

attendant answers.
5

display.

attended

attendant answers.
6
transfer attended.

display.

9.9 FAX
Test
Id
ONT Fax
1
pages

pages
Test
Id

1 system feature

via PSTN loopback

G729
2 G729 Not supported
via PSTN loopback

G723
via PSTN loopback

Test
Id
1
Same test as 6.9.2.
pages
9.10 Modem VBD

Test
Id
1
For reliable modem transport,
the system needs a
composite clock input to the
OLT.

Test
Id
1 Check that existing calls Not Tested
Not Tested
3 Check that an existing call can be Not Tested
ONT reboot
Check that calls are possible as soon
Remote reboot ONT and then
as ONT has come back to service.
6 measure time to dial tone.
729 Pass (3:35s)
Temporary IP Link down with the dial tone, then fast busy on
OXE dial complete. After
Check that the ?? tone is presented registration timeout
7 to the user immediate fast busy.
Check that incoming/outgoing calls Took several minutes after
are possible once the link is recovery for the PBX to begin
reestablished. forwarding invites to lines.

10 Test Results - 702
10.1 Commissioning
Test
Id
Address
Configure all your SIP sets from MCDU
number SIPset-1 to SIPset-8
Check the registration on your sets and assignment.
the display

matter.
2 DNS IP address <Secondary_DNS_IP>.
name : <DNS_Name>.

display.
DHCP registration
3
NTP registration
to retrieve the date and time from the
OXE Node 1 main IP address
to ONT via ME, verified
4 <Node1_Main_IP@>.
displays it.
Brief” (1) By default,
The phone SIPset-1 is configured with a OXE>SIP>SIP Registrar>Min
5 value lower than 1800 seconds. Expires=1800s
702 Pass Requires Reboot to
Check the phone registration and change.
display.

authentication
system.

6
display.

rejected.
Signalling TCP-UDP.
7 over TCP
In the two cases, check the registration,

and basic calls.
Notes:

framing).
Phones:
OXE :
Sip or rtp
Test
Id
domain)
1
Check audio quality

(extra-domain)
Check that the call is established G729 not supported
2
Check audio quality

order
MOS of 4.
domain)
Check audio quality


Check audio quality

Redo the same tests


(extra-domain)
5
Check audio quality

Redo the same tests

up phase.
Test
Id
Local call

1 answer the call.

but,

SIPset-2.

T2 Loopback)
shown correctly
3 SIPset-1)


non-ASCII characters. Check caller Switch won’t accept the
display. characters at Telnet interface,
With SIPset-2 call SIPset-1 (extension can’t test. I have verified with
4 with a name containing non-ASCII following test (#5) that lines
characters). send whatever is in the Display
name. Will work on all ONTs.

Test
Id

output matches display name
5
and is properly formatted. (i.e.
verified ONT is correct in all
cases, problem is phones)


OXE configuration)
ID secret: True

hidden. ONT retrieves CLIR information
from “anonymous” in userinfo
Do the same test but by calling an of SIP From header and displays
6 external PSTN number. “Unknown Number”.


NOT hidden.

Can’t figure out how to get it to
With SIPset-2 call SIPset-1 by dialing recognize the per call caller ID
*31*SIPset-1# (dial *31* + <target code. This is switch function,
7 MCDU number>#) in order to hide just tests dial plan, so just
SIPset-2 identity. captured to show dial plan
conformance.
identity is hidden.

Test
Id


ok.

Mail
Call waiting

the call.
SIPset-3.
Check that the Call Waiting tone is

served to SIPset-2. hook) and Recall button
9
behavior supported.
“R”+SOC 1
the waiting call

Call to busy user

the call.
With SIPset-3 call SIPset-2 and wait to ONT returns a 486 Busy Here
make it busy message. Ignore 404 not found
10 in trace, switch had lingering
With SIPset-4 call SIPset-2 which is busy registration for 1107 that hadn’t
timed out.

call.
11

Test
Id
OXE

DND
We are not aware of the call
12 forwarding state so there is no
distinctive dial tone.
OXE.

forward (CFU).
OXE.
On our switch Immediate
13 Forward is *60, clear is *64.
Switch side feature.
hung up.

call.

cancellation.
reply (CFNR).
back tone from OXE. *62 is CFNA
When on, forwards, when off,
cancel.
sec).


Test
Id

call.

cancellation.
(CFB).
back tone from OXE.

call.
disabled on Dynamic State for
15 make SIPset-1 busy
this to work.

the call.

call.

cancellation.
Call Pick Up.
17
Distinctive ringing
SIPset-1
18
1,
SIPset-1
19 Config note: must use only
phone # in config, not full sip uri

Test
Id
Check that automatic call is established for 729. Fixed in future release.

Check that SIPset-1 can dial some 702 Not supported, hotline only.
Check if call is maintained or released Clarified that Re-Invite is
after the session timer has expired: acceptable.
With SIPset-1 dial SIPset-2. Re-invite only initiated by
switch. Switch will terminate
Check that call is maintained or With Re-invite that is all you
release. can do.

Unplug the SIPset-2
If line is registered but
From SIPset-1, call SIPset-2 while unresponsive forward to voice
SIPset-2 still registered mail after 3 failed invites.
Subsequent calls go
Check the ring back tone and display immediately to voice mail. This
22
case works.
From SIPset-1, call SIPset-2 once SIPset- If I unregister as shown in the
nd
2 not registered anymore 2 part of the use case lines
SIP: “480 Temporarily Unavailable” will will give reorder/fast busy.
be sent 729GP gives regular busy.

23

While I cannot cause this
sequence due to not being able
Call rejected by call handling to configure the switch to do
(SIP: “183 Progress/487 Request this. This sequence can be
24
Terminated”) seen in lots of call sequences
e.g. max number of calls reached etc. and the lines all respond
properly and are operational
after the Call rejection via 487.

Test
Id

From SIPset-1 put SIPset-2 on hold North American behavior

1 with “R” key. (flash hook) and Recall button
Check dial tone and dial SIPset-3 behavior supported.

display.

conversation.

North American behavior
2 (flash hook) and Recall button
calls
behavior supported.
Call park:
call.
MCDU number>)
Check that OXEset-2 is put on hold. North American behavior
3 (flash hook) and Recall button
Release SIPset-1 but not SIPset-2. behavior supported.

number>).
Send/receive DTMF
using RFC 2833
4
From SIPset-1 call the Voice Mail at

Test
Id
menus
Three party conference
After enquiry call, dial “R” + 3 key to

activate three party conference.

For North America, it is A calls
B, flash B calls C flash, 3 way
call is activated.
5 conference. Digits after the
Do the test with all 3 participants flash do not control legs of
being SIPsets. conference.



10.5 Call Transfer




C.
(Section 4.5)

the call.

Case
Id
A B C


connected.

Case
Id
A B C



10.6 Voice Mail
guide)
Test
Id
14999.

Hook.

blinking)
14999.
2
it and remove it.

removal.

hangup.
3
rejection.

service access.

Mail
Mail number>).

Voice Mail.

Voice mail deposit
5

10.7.1 Provisioning
Test
Id
room set

Configured it as “Room &
Or
1 Normal”

OXE.
administrative set

set Configured it as
Or “Administrative & Normal”
2
administrative set

OXE.
booth set

is declared as a hotel booth set Configured it as “Home &
Or Normal”
3

OXE.

Test
Id

Or

2 parameter set.
information.
3
the SIP phone room

10.7.3 Voice mail
Voicemail number.
Test
Id



10.7.4 Wake up
Test
Id
SIP phone

1
room SIP phone

2
phone set.
suite SIP phone

the prefix 584
3

suite SIP phone

the prefix 506
4

5
suite SIP phone

Test
Id

prefix 585


Test
Id
room SIP phone set.

1 room status).
problem).
10.7.6 Mini-Bar
Test
Id

OXE.

Test
Id

the OXE

number

established.
phone number

established.
phone number

established.

established.

established.

Test
Id
1

2
time)

3

10.8 Attendant
are tested.
Test
Id
1 answers.

audio.
attendant.

2

properly managed.
attended

answers.


display.

attended

4 answers.

transfer attended.

display.

semi-attended

attendant answers.
5

display.

attended

attendant answers.
6
transfer attended.

display.

10.9 FAX
Test
Id
ONT Fax
1
pages

pages
Test
Id

1 system feature

via PSTN loopback

G729
via PSTN loopback

G723
via PSTN loopback

Test
Id
1
Same test as 6.9.2.
pages
10.10 Modem VBD

Test
Id
1
For reliable modem
transport, the system needs
a composite clock input to
the OLT.

Test
Id
1 Check that existing calls
3 Check that an existing call can be
ONT reboot
Check that calls are possible as soon
Remote reboot ONT and then
as ONT has come back to service.
6 measure time to dial tone.
702 Pass (1:06s)
dial tone, then fast busy on
Temporary IP Link down with the
dial complete. After
OXE
registration timeout
Check that the ?? tone is presented
immediate fast busy.
7 to the user
Took several minutes after
Check that incoming/outgoing calls
recovery for the PBX to begin
are possible once the link is
forwarding invites to lines.
reestablished.
Not sure why.

11 Test Results - 704
11.1 Commissioning
Test
Id
Address
Check the registration on your sets assignment.
and the display

matter.
DNS IP address
2
<Secondary_DNS_IP>.
name : <DNS_Name>.

display.
DHCP registration
3
NTP registration
to ONT via ME, verified
4 <Node1_Main_IP@>.
displays it.
Support of “423 Interval Too By default,
Brief” (1) OXE>SIP>SIP Registrar>Min
The phone SIPset-1 is configured with Expires=1800s
5 a value lower than 1800 seconds.
704 Fail Required Reboot,
Check the phone registration and Failed to renegotiate.
display. 729GP Pass

authentication
system.

6
display.

rejected.
Signalling TCP-UDP.
UDP Supported by 704.
704 does not support TCP or
7 over TCP
TLS.
Notes:

framing).
Phones:
OXE :
Sip or rtp
Test
Id
domain)
Check that the call is established 704 Fail sends uLaw.
1
Check audio quality

(extra-domain) g.729 is not supported. Low
Check that the call is established bw codecs not needed,
2
using direct RTP in G729. system is GigE at all points.
Check audio quality

order
MOS of 4.
domain)
Check audio quality


Check audio quality

Redo the same tests


(extra-domain)
5
Check audio quality

Redo the same tests

up phase.
Test
Id
Local call

1 answer the call.

but,

SIPset-2.

T2 Loopback)
shown correctly
3 SIPset-1)


non-ASCII characters. Check caller Switch won’t accept the
display. characters at Telnet interface,
With SIPset-2 call SIPset-1 (extension can’t test. I have verified with
4 with a name containing non-ASCII following test (#5) that lines
characters). send whatever is in the Display
name. Will work on all ONTs.

Test
Id

output matches display name
5
and is properly formatted. (i.e.
verified ONT is correct in all
cases, problem is phones)


OXE configuration)
ID secret: True

call and check that SIPset-2 identity is ONT retrieves CLIR information
hidden. from “anonymous” in userinfo
of SIP From header and displays
Do the same test but by calling an “Unknown Number”.
704 Pass says “Private.

NOT hidden.

Can’t figure out how to get it to
With SIPset-2 call SIPset-1 by dialing recognize the per call caller ID
*31*SIPset-1# (dial *31* + <target code. This is switch function,
7 MCDU number>#) in order to hide just tests dial plan, so just
SIPset-2 identity. captured to show dial plan
conformance.
identity is hidden.

Test
Id


ok.

Mail
Call waiting

the call.
SIPset-3. Used N. American method.
Recall button method only
Check that the Call Waiting tone is supported by 702 and 729.
served to SIPset-2.
9
Call Waiting is provided by the
ONT.
“R”+SOC 1
the waiting call

Call to busy user

the call. ONT returns a 486 Busy Here
With SIPset-3 call SIPset-2 and wait to message. Ignore 404 not found
make it busy in trace, switch had lingering
10 registration for 1107 that hadn’t
With SIPset-4 call SIPset-2 which is busy timed out.

call.
11

Test
Id
OXE

DND
We are not aware of the call
12 forwarding state so there is no
distinctive dial tone.
OXE.

forward (CFU).
OXE. On our switch Immediate
Forward is *60, clear is *64.
13
With SIPset-3 call the SIPset-1. Switch side feature.
hung up.

call.

cancellation.
reply (CFNR).
CFNR. Wait for acknowledgement ring *62 is CFNA
back tone from OXE. When on, forwards, when off,
14 With SIPset-3 call the SIPset-1. Check cancel.
sec).


Test
Id

call.

cancellation.
(CFB).
back tone from OXE.

call.
disabled on Dynamic State for
15 make SIPset-1 busy
this to work.

the call.

call.

cancellation.
Call Pick Up.
17
Distinctive ringing
SIPset-1
18
1,
SIPset-1
19

Test
Id

Check if call is maintained or released Clarified that Re-Invite is
after the session timer has expired: acceptable.
With SIPset-1 dial SIPset-2. Re-invite only initiated by
switch. Switch will terminate
Check that call is maintained or With Re-invite that is all you
release. can do.

Unplug the SIPset-2
From SIPset-1, call SIPset-2 while If line is registered but

SIPset-2 still registered unresponsive forward to voice
mail after 3 failed invites.
Check the ring back tone and display Subsequent calls go
22
immediately to voice mail. This
From SIPset-1, call SIPset-2 once SIPset- case works.
2 not registered anymore If I unregister as shown in the
nd
SIP: “480 Temporarily Unavailable” will 2 part of the use case lines
be sent will give reorder/fast busy.

23

While I cannot cause this
sequence due to not being able
Call rejected by call handling to configure the switch to do
(SIP: “183 Progress/487 Request this. This sequence can be
24
Terminated”) seen in lots of call sequences
e.g. max number of calls reached etc. and the lines all respond
properly and are operational
after the Call rejection via 487.


Test
Id

From SIPset-1 put SIPset-2 on hold Only North American

1 with “R” key. flashook behavior supported
Check dial tone and dial SIPset-3 on the 704.

display.

conversation.

Only North American
2 flashook behavior supported
calls
on the 704.
Call park:
call.
MCDU number>)
Call prefix on my switch is
Check that OXEset-2 is put on hold.
*75.
3
Release SIPset-1 but not SIPset-2.

number>).
Send/receive DTMF
using RFC 2833
4 From SIPset-1 call the Voice Mail at

menus
Three party conference For North America, it is A
5 calls B, flash B calls C flash, 3
After enquiry call, dial “R” + 3 key to way conference. Digits after

Test
Id
activate three party conference. the flash do not control legs
of conference.

call is activated.

being SIPsets.



11.5 Call Transfer




C.
(Section 4.5)

the call.

Case
Id
A B C


connected.

Case
Id
A B C



11.6 Voice Mail
guide)
Test
Id
14999.
704 Does not support
1 Check that the MWI (Audible Tone) is subscribe so no MWI.
Hook.

blinking)
14999.
704 Does not support
2 subscribe.
it and remove it.

removal.

hangup.
3
rejection.

service access.

Mail
Mail number>).

Voice Mail.

Voice mail deposit
5

11.7.1 Provisioning
Test
Id
room set

Configured it as “Room &
Or
1 Normal”

OXE.
administrative set

set Configured it as
Or “Administrative & Normal”
2
administrative set

OXE.
booth set

is declared as a hotel booth set Configured it as “Home &
Or Normal”
3

OXE.

Test
Id

Or

2 parameter set.
information.
3
the SIP phone room

11.7.3 Voice mail
Voicemail number.
Test
Id

704 N/A (unsolicited
NOTIFY prob, msg works,
no MWI)

704 N/A (unsolicited
NOTIFY prob, msg works,
no MWI)

11.7.4 Wake up
Test
Id
SIP phone

1
room SIP phone

2
phone set.
suite SIP phone

the prefix 584
3

suite SIP phone

the prefix 506
4

5
suite SIP phone

Test
Id

prefix 585


Test
Id
room SIP phone set.

1 room status).
problem).
11.7.6 Mini-Bar
Test
Id

OXE.

Test
Id

the OXE

number

established.
phone number

established.
phone number

established.

established.

established.

Test
Id
1

2
time)

3

11.8 Attendant
are tested.
Test
Id
1 answers.

audio.
attendant.

2

properly managed.
attended

answers.


display.

attended

4 answers.

transfer attended.

display.

semi-attended

attendant answers.
5

display.

attended

attendant answers.
6
transfer attended.

display.

11.9 FAX
Test
Id
ONT Fax
1
pages

pages
Test
Id

1 system feature

via PSTN loopback

G729
2 G729 Not supported
via PSTN loopback

G723
via PSTN loopback

Test
Id
1
Same test as 6.9.2.
pages
11.10 Modem VBD

Test
Id
1 in Passthrough mode available in most locations.
For reliable modem transport,
the system needs a composite
clock input to the OLT.

Test
Id
1 Check that existing calls
3 Check that an existing call can be
ONT reboot
Check that calls are possible as soon Remote reboot ONT and then
as ONT has come back to service. measure time to dial tone.
6
704 Pass (1:12s)
Temporary IP Link down with the dial tone, then fast busy on
OXE dial complete. After
Check that the ?? tone is presented registration timeout immediate
7 to the user fast busy.
Check that incoming/outgoing calls Took several minutes after
are possible once the link is recovery for the PBX to begin
reestablished. forwarding invites to lines. Not
sure why.

12 Appendix A: ONT Configuration requirements
Equipment Settings profile for use with all ONTs.

Services Profile Settings for 702, 704, 705. MLPP should be off.

729GP Services Profile. Only the Outbound proxy differs.

Call Features Profile for all ONTs :

Call Features Profile 2nd Half. All below Message Waiting is defaults.

How to configure Hotline in the Call Features Profile.

Example of Per Line Configuration :

Example of In service Line, Activate/Deactivate buttons used to put line into/out of service:
Call State shows line offhook state.

Registration State show whether line is registered and any problems with registration.

13 Appendix B: OmniPCX Enterprise
configuration requirements
13.1 Global Parameters

Sip Trunk Configuration:
- Trunk Group Type: T2
- Remote Network: 14
- Q931 Signal variant: ABC-F
- Number of digits to send: 0
- T2 Specification: SIP
- Public Network COS: 31
- IP Compression Type: G711
Sip Gateway Configuration:

- Sip Subnetwork: 14 (<- Remote Network used above)
- Sip Trunk Group: XX (<- Trunk group used above)
Since Tellabs ONTs only support G.711 codec, all IP domains are setup setup Without Compression.
13.2 Configuration of the SIP users

All analog phones connected to ONT’s are setup as SIP EXTENSION in the OXE.
13.3 Prefix Plan

Feature Code OXE Prefix Feature at user level
*21* Immediate_forward Call Forwarding
Unconditional (CFU)
activation *
#21# Forward_cancellation Call Forwarding
Unconditional (CFU)
deactivation
*61* Forward_on_no_reply Call Forwarding No Reply
(CFNR)
activation *
#61# Forward_cancellation Call Forwarding No Reply
(CFNR)
deactivation
*67* Immediate_forward_on_busy Call Forwarding Busy (CFB)
activation *
#67# Forward_cancellation Call Forwarding Busy (CFB)

deactivation
*31* Secret / Identity CLIR per call
* to a user or to the voice mail

** for cancelling any kind of Call Forward

13.4 Suffix Plan
1 - Broker Call
2 - Consultation Call
3 - Three-Party Conference
4 - Barge-in (Intrusion) => not applicable
5 - Callback On Free Or Busy Set => applicable ?
6 - Busy Camp-on => not applicable
7 - Paging Request => not applicable
8 - Voice Mail Deposit => applicable ?
* - DTMF end-to-end dialing => not applicable
13.5 Tones played by OXE

Describe the tones played by the OXE and their configuration to be able to change that
configuration for consistency with the ONT.

14 Appendix C: AAPP member’s escalation
process
Contact name TELLABS TAC SUPPORT

Title 24x7 Technical Support
Address 1415 WEST DIEHL ROAD

City NAPERVILLE, IL
ZIP 06825
Country USA
Phone +1.800.443.5555
WEB http://www.tellabs.com

15 Appendix D: AAPP program
15.1 Alcatel-Lucent Application Partner Program (AAPP)
The Application Partner Program is designed to support companies that develop communication
applications for the enterprise market, based on Alcatel-Lucent Enterprise's product family.
The program provides tools and support for developing, verifying and promoting compliant third-
party applications that complement Alcatel-Lucent Enterprise's product family. ALE International
facilitates market access for compliant applications.
The Alcatel-Lucent Application Partner Program (AAPP) has two main objectives:
 Provide easy interfacing for Alcatel-Lucent Enterprise communication products:

Alcatel-Lucent Enterprise's communication products for the enterprise market include
infrastructure elements, platforms and software suites. To ensure easy integration, the
AAPP provides a full array of standards-based application programming interfaces and
fully-documented proprietary interfaces. Together, these enable third-party applications to
benefit fully from the potential of Alcatel-Lucent Enterprise products.
 Test and verify a comprehensive range of third-party applications:

to ensure proper inter-working, ALE International tests and verifies selected third-party
applications that complement its portfolio. Successful candidates, which are labelled
Alcatel-Lucent Enterprise Compliant Application, come from every area of voice and data
communications.
The Alcatel-Lucent Application Partner Program covers a wide array of third-party

applications/products designed for voice-centric and data-centric networks in the enterprise market,
including terminals, communication applications, mobility, management, security, etc.

Web site
The Application Partner Portal is a website dedicated to the AAPP program and where the
InterWorking Reports can be consulted. Its access is free at
http://applicationpartner.alcatel-lucent.com
15.2 Enterprise.Alcatel-Lucent.com
You can access the Alcatel-Lucent Enterprise website at this URL: http://www.enterprise.alcatel-
lucent.com/

16 Appendix E: AAPP Escalation process
16.1 Introduction
The purpose of this appendix is to define the escalation process to be applied by the ALE
International Business Partners when facing a problem with the solution certified in this document.
The principle is that ALE International Technical Support will be subject to the existence of a valid
InterWorking Report within the limits defined in the chapter “Limits of the Technical support”.
In case technical support is granted, ALE International and the Application Partner, are engaged as
following:
(*) The Application Partner Business Partner can be a Third-Party company or the ALE
International Business Partner itself

16.2 Escalation in case of a valid Inter-Working Report
The InterWorking Report describes the test cases which have been performed, the conditions of the
testing and the observed limitations.
This defines the scope of what has been certified.
If the issue is in the scope of the IWR, both parties, ALE International and the Application Partner,
are engaged:
Case 1: the responsibility can be established 100% on ALE International side.

In that case, the problem must be escalated by the ALE Business Partner to the ALE
International Support Center using the standard process: open a ticket (eService Request –
eSR)
Case 2: the responsibility can be established 100% on Application Partner side.

In that case, the problem must be escalated directly to the Application Partner by opening a
ticket through the Partner Hotline. In general, the process to be applied for the Application
Partner is described in the IWR.
Case 3: the responsibility can not be established.

In that case the following process applies:
 The Application Partner shall be contacted first by the Business Partner (responsible for
the application, see figure in previous page) for an analysis of the problem.
 The ALE International Business Partner will escalate the problem to the ALE
International Support Center only if the Application Partner has demonstrated with
traces a problem on the ALE International side or if the Application Partner (not the
Business Partner) needs the involvement of ALE International
In that case, the ALE International Business Partner must provide the reference of the Case
Number on the Application Partner side. The Application Partner must provide to ALE
International the results of its investigations, traces, etc, related to this Case Number.
ALE International reserves the right to close the case opened on his side if the
investigations made on the Application Partner side are insufficient or do not exist.
Note: Known problems or remarks mentioned in the IWR will not be taken into account.
For any issue reported by a Business Partner outside the scope of the IWR, ALE International offers
the “On Demand Diagnostic” service where ALE International will provide 8 hours assistance
against payment .
IMPORTANT NOTE 1: The possibility to configure the Alcatel-Lucent Enterprise PBX with ACTIS
quotation tool in order to interwork with an external application is not
the guarantee of the availability and the support of the solution. The reference remains the
existence of a valid InterWorking Report.
Please check the availability of the Inter-Working Report on the AAPP (URL:
https://applicationpartner.alcatel-lucent.com) or Enterprise Business Portal (Url: Enterprise Business
Portal) web sites.
IMPORTANT NOTE 2: Involvement of the ALE International Business Partner is mandatory, the
access to the Alcatel-Lucent Enterprise platform (remote access, login/password) being the
Business Partner responsibility.

16.3 Escalation in all other cases
For non-certified AAPP applications, no valid InterWorking Report is available and the integrator is
expected to troubleshoot the issue. If the ALE Business Partner finds out the reported issue is
maybe due to one of the Alcatel-Lucent Enterprise solutions, the ALE Business Partner opens a
ticket with ALE International Support and shares all trouble shooting information and conclusions
that shows a need for ALE International to analyze.
Access to technical support requires a valid ALE maintenance contract and the most recent
maintenance software revision deployed on site. The resolution of those non-AAPP solutions cases
is based on best effort and there is no commitment to fix or enhance the licensed Alcatel-Lucent
Enterprise software.
For information, for non-certified AAPP applications and if the ALE Business Partner is not able to
find out the issues, ALE International offers an “On Demand Diagnostic” service where assistance
will be provided for a fee.

16.4 Technical support access
The ALE International Support Center is open 24 hours a day; 7 days a week:
 e-Support from the Application Partner Web site (if registered Alcatel-Lucent Application
Partner): http://applicationpartner.alcatel-lucent.com
 e-Support from the ALE International Business Partners Web site (if registered Alcatel-Lucent
Enterprise Business Partners): https://businessportal2.alcatel-lucent.com click under “Contact
us” the eService Request link
 e-mail: Ebg_Global_Supportcenter@al-enterprise.com
 Fax number: +33(0)3 69 20 85 85
 Telephone numbers:
ALE International Business Partners Support Center for countries:
Country Supported language Toll free number

France
Belgium French
Luxembourg
Germany
Austria German
Switzerland
United Kingdom
Italy
Australia
Denmark
Ireland
Netherlands +800-00200100
South Africa
Norway
English
Poland
Sweden
Czech Republic
Estonia
Finland
Greece
Slovakia
Portugal
Spain Spanish
For other countries:

English answer: + 1 650 385 2193
French answer: + 1 650 385 2196
German answer: + 1 650 385 2197
Spanish answer: + 1 650 385 2198
END OF DOCUMENT


Iwr 0200 Ed01 - Tellabs Ont - Omnipcxenterpriser11.2 PDF

Caricato da

Informazioni sul documento

Titolo originale

Copyright

Formati disponibili

Condividi questo documento

Condividi o incorpora il documento

Opzioni di condivisione

Hai trovato utile questo documento?

Questo contenuto è inappropriato?

Copyright:

Formati disponibili

Iwr 0200 Ed01 - Tellabs Ont - Omnipcxenterpriser11.2 PDF

Caricato da

Copyright:

Formati disponibili

ALE Application Partner Program

ALE INTERNATIONAL MAKES NO REPRESENTATIONS, WARRANTIES OR CONDITIONS WITH RESPECT TO THE

ALE Application Partner Program – Inter-working report - Edition 1 - page 1/153

Date of the certification March 2016

ALE International representative Frank Gadot

Alcatel-Lucent Enterprise OmniPCX Enterprise

Edition 1: creation of the document – March 2016

Passed with restrictions

Refer to the section 0 for a summary of the test results.

IWR validity extension

ALE Application Partner Program – Inter-working report - Edition 1 - page 2/153

Contact name TELLABS TAC SUPPORT

Address 1415 WEST DIEHL ROAD

ALE Application Partner Program – Inter-working report - Edition 1 - page 3/153

ALE Application Partner Program – Inter-working report - Edition 1 - page 4/153

ALE Application Partner Program – Inter-working report - Edition 1 - page 5/153

ALE Application Partner Program – Inter-working report - Edition 1 - page 6/153

It certifies proper inter-working with the AAPP member’s application.

- the Technical Support page of the Enterprise Business Portal

- the Application Partner portal (https://applicationpartner.alcatel-lucent.com) with free

ALE Application Partner Program – Inter-working report - Edition 1 - page 7/153

A new release is identified as following:

ALE Application Partner Program – Inter-working report - Edition 1 - page 8/153

3.1 Case of additional Third party applications

ALE Application Partner Program – Inter-working report - Edition 1 - page 9/153

Application commercial name: GPON ONT 705GR

Application version: version: FP25.7.1

Most switch feature activation/deactivation is performed by using Vertical Service Codes or

Brief application description:

ALE Application Partner Program – Inter-working report - Edition 1 - page 10/153

5.1 General architecture

ALE Application Partner Program – Inter-working report - Edition 1 - page 11/153

ALE Application Partner Program – Inter-working report - Edition 1 - page 12/153

5.3 Software configuration

ALE Application Partner Program – Inter-working report - Edition 1 - page 13/153

6.1 Summary of main functions supported

705GR 702 729GP 704 Comment

ALE Application Partner Program – Inter-working report - Edition 1 - page 14/153

6.2 Summary of problems

6.3 Summary of limitations

 For this certification only one OXE CPU was used.

 G.711 is the only codec supported.

 TLS is not supported by Alcatel-Lucent OXE R11.2 for SIP communication

 ONT 704 cannot support g.711 aLaw.

 704 does not support DNS.

6.4 Notes, remarks

ALE Application Partner Program – Inter-working report - Edition 1 - page 15/153

Test case 2 The application

ALE Application Partner Program – Inter-working report - Edition 1 - page 16/153

Note that authentication is disabled

Check the phone registration and

ALE Application Partner Program – Inter-working report - Edition 1 - page 17/153

Configure the phone SIPset-1 with

Redo the same tests with a wrong

In the two cases, check the

ALE Application Partner Program – Inter-working report - Edition 1 - page 18/153

Note: ONT and OXE will use 20ms framing.

Call from SIPset-1 to SIPset-3

Set system law = mu-law

Configure SIPset-1 to use VAD

Call from SIPset-1 to SIPset-2 (intra-

ALE Application Partner Program – Inter-working report - Edition 1 - page 19/153

Redo the same tests