Ascom IP-DeCT Handset SIP Verification Final Test Report

tekVizion Confidential
SIP Verification Test Plan and

Report for Ascom IP-DECT
System
SIP V Test Result PASS
Test Date 04/23/2008
Product Name Ascom 9d24 DECT Handset
Product Version # (must be IPBS[2.3.5]
generally available)
Unified Communications 6.1.1.2000-3
Manager Version X.X(x)
Cisco Security Agent Version csa-5.0.0-216.1
(CSA)
Product Type (Billing, Voice IP-DECT Phone
Recording, phone apps etc):
API/Protocol(s) Used SIP
SIP V Partner Contact Name: Staffan rnbratt
SIP V Partner Contact Phone: +46 31 55 93 29
SIP V Partner Contact Email: Staffan.Ornbratt@ascom.se
SIP V Lab Location (US): US
Partner Main Support Number
Partner Main Support Email
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Revision History
Revision Author Date Comment
0.1 Gerry Pearson 8/15/2006 Initial draft
0.2 Abelardo Guajardo 9/7/2006 DTMF modifications
0.3 Srikanth Palla 04.24/2008 Updated with test results
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Table of Contents
1 Introduction .............................................................................................................. 4
1.1 Document Scope ................................................................................................ 4
1.2 tekVizion Labs ................................................................................................... 4
1.3 tekVizion PVS, Inc............................................................................................. 6
2 Testing Process and Criteria...................................................................................... 8
2.1 Test Approach .................................................................................................... 8
2.2 Entry Criteria ..................................................................................................... 8
2.3 Exit Criteria........................................................................................................ 8
3 Product Overview ................................................................................................... 11
4 Executive Summary ................................................................................................ 12
5 Summary of Testing Results.................................................................................... 13
5.1 Features Tested for Interoperability to the Cisco Unified Communications
Manager.................................................................................................................... 13
5.2 Interpretation of Test Results............................................................................ 14
5.3 Installation, Functional, Negative, and Informational Test Summary ................ 14
5.4 Performance Test Summary ........................................................................ 14
5.5 Items Not Tested .............................................................................................. 15
5.6 Assumptions..................................................................................................... 15
6 Test Environment.................................................................................................... 16
6.1 Administration, Testing and Debugging tools................................................... 16
6.2 Equipment Requirements ................................................................................. 16
6.3 3rd Party Supporting Documentation and Urls........................................... 17
6.4 Test Environment Architecture ........................................................................ 19
7 Test Cases ............................................................................................................... 20
7.1 Basic Call Scenarios......................................................................................... 20
7.2 Cisco Unified Communications Manager Standard SIP Feature Support........... 24
7.3 System Control and Verification....................................................................... 66
7.4 SIP Endpoint Functionality Verification ........................................................... 68
7.5 Support for SIP-based RFCs and Drafts............................................................ 69
7.6 Reliability and Stability.................................................................................... 71
7.7 SIP Torture Tests.............................................................................................. 74
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1 Introduction
This document is the detailed Interoperability Verification Test Plan and Report for
Cisco Unified Communications Manager 6.1.1 and IP-DECT Base Station,
Ascom 9d24 DECT Handset. SIP Verification Testing for 3rd party endpoints and
Ciscos Unified Communications Manager is performed on-site at tekVizion Labs.
SIP Verification Testing tests a single Ciscos Unified Communications Manager
version to another single endpoint vendors specific product version. Verification
testing uses tekVizion Labs SIP Verification Test Plan.
When two products have successfully passed tekVizions SIP Verification testing with
no outstanding issues, the particular versions of the products are verified as
interoperable. The Verification only applies to the product versions used in the test,
and are only verified as interoperable in a configuration matching that under which
the test took place, and for the features and functions tested as part of the test plan.
If the testing was successful, a SIP Verification Logo will be provided.
1.1 Document Scope

This document provides the test strategy and approach used by tekVizion Labs to
perform SIP Verification between two products. Its purpose is to provide prospective
customers with a high-level overview of the general criteria, process and scope of
testing that will be performed. The complete test strategy will be available for review,
upon request.
1.2 tekVizion Labs

tekVizion Labs (www.tekvizionlabs.com) is an independent testing and certification
facility offered by tekVizion PVS, Inc. (tekVizion). tekVizion Labs offers third party
interoperability certification of VoIP products and applications and it addresses the
growing demand for interoperability testing in the VoIP Industry. Since inception,
tekVizion engineers have been integrating and testing multi-vendor solutions for
service providers, making tekVizion uniquely qualified to perform certification testing.
Leveraging this extensive interoperability test experience and hands-on expertise
with more than 150 next-generation products, combined with tekVizions experience
integrating and testing multi-vendor solutions for service providers, tekVizion is able
to streamline the process leading to successful and repeatable product certification
testing.
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tekVizion Labs offers several services including but not limited to the following:
tekVizion Certification Testing

tekVizion offers product interoperability certification of one product to another and
offers multi-vendor solution certification. Product Certification testing is
performed on-site at tekVizion Labs by tekVizion engineers. Certification
Testing tests a single vendor's specific product version to another single
vendors specific product version. Certification testing uses tekVizion Labs
Standard Test Plan. Upon completion, tekVizion Labs will deliver a final
test report. If the testing is successful, a Statement of Certification will be provided.
Pricing for Certification Testing is provided as a daily rate and the number of days
varies based on the types of products being tested.
tekVizion SIP Verification Testing

tekVizion Labs is the authorized test facility for SIP Verification testing between
Ciscos Unified Communications Manager and third party SIP end points.
When two products have successfully passed tekVizions SIP Verification
testing with no outstanding issues, the particular versions of the products
are verified as interoperable. The Verification only applies to the product
versions used in the test and if the testing was successful, a Statement of
Verification and SIP Verification Logo will be provided.
Third-Party Certification/Validation Testing

Many companies or organizations have already implemented their own certification
or validation programs with their own test plans, procedures or processes yet need
resources or additional lab space to implement or maintain them. tekVizion Labs is
equipped to implement this type of testing on behalf of another company or
organization. In many cases, we can build or assist in building a new program from
the ground up.
Solution Testing
Solution Testing of a specific multi-vendor configuration can also be provided by
tekVizion Labs. This can be performed within tekVizion Labs or at another
companys testing facility. tekVizion Solution Certification can be achieved when
tekVizion Labs test plans are used and the solution passes all test criteria. Pricing for
solution testing or solution certification is provided as a custom quote. tekVizion Labs
makes this service available to vendors as well as service providers who do not
currently have their own lab in which to test their solutions prior to deploying into a
live network.
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Remote Testing
tekVizion offers secure remote access to certain products in our lab for a specified
amount of testing time. A fixed number of consulting hours are also provided as part
of this service based upon the remote access time contracted. Pricing is provided as
a fixed fee for the initial time requested. Additional testing time and consulting hours
can be provided as needed for an additional fee. tekVizion Labs can also perform
remote testing to products in another companys lab.
PBX/IP PBX Trunk Interface Interoperability Testing

tekVizion Labs has a variety of PBX and IP PBX systems available for testing. With
these systems, we provide interoperability testing of trunking interfaces between
legacy or IP PBXs and VoIP networks
IMS Conformance / Interoperability Testing

tekVizion Labs can verify products designed for IMS networks including testing to the
IMS specifications, testing interoperability among IMS components and certification
of third-party applications into trusted IMS networks. For example, tekVizion Labs
can provide ISC Interface Certification to certify the interoperability of a specific AS to
SCIMs/S-CSCFs, SCIMs to S-CSCFs, etc.
Product/Compliance Assessment
Leveraging extensive interoperability test experience, broad knowledge and hands-
on expertise of over 150 next-generation products, tekVizion can provide vendors
with an objective product assessment. A product assessment may include
verification of product functionality, interface usability, assessment of differentiating
features, stress and performance testing, and protocol conformance and
implementation. Compliance assessments test a product for compliance with a
particular industry specification.
Open Testing
In addition to the categories listed above, tekVizion Labs can help with just about any
type of testing including testing against customer-provided test plans, system level
testing, systems integration or regression testing.
1.3 tekVizion PVS, Inc.

tekVizion has set the new standard for systems integration in the
telecommunications industry. As long-time telecom veterans, we are committed to
helping service providers speed up the introduction of new, innovative and integrated
services.
tekVizions core services include

Consulting
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Custom applications
Integration and support
Testing through tekVizion Labs
tekVizion helps service providers achieve a smooth transition to packet-voice

networks, speeding delivery of integrated services. Leveraging real-world, multi-
vendor integration and test experience and proven processes, tekVizion provides
solutions that meld legacy and next-generation resources.
The tekVizion team brings together experience from leading vendors and service
providers in telecom. Our unique expertise includes unparalleled experience on a
vast array of VoIP, IMS and other next-generation products and technologies, as well
as, legacy switching services and platforms. We rely on this combined experience to
do what we do best: help our clients leverage their existing assets, while introducing
new network elements and applications to advance the rollout of services that excite
customers and result in new revenues for the bottom line.
Leveraging this experience and following our proven processes and methodologies,
tekVizion offers services to service providers, large enterprises, product vendors and
other professional services firms.
(For more information on tekVizion and its practice areas, please visit tekVizions
web site at www.tekVizion.com.)
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2 Testing Process and Criteria
2.1 Test Approach

Black box testing methodology will be used for SIP Verification testing. The testing
approach is to test the functional interoperability between the products under test. In
certain instances where known interoperability issues exist between two
components, the test team will specifically test for the known problems. The devices
under test will have to pass all the test cases to be verified.
In the event that a vendors product does not support a specific feature or function, it
will not be tested and recorded as N/S (not supported) in the test report. Features
that are not supported may cause consideration for failure from a SIP Verification
perspective.
2.2 Entry Criteria

Before verification testing can be scheduled, the parties contracting the service must
sign a Testing Services Agreement and resolve appropriate financial obligations as
outlined in the Agreement. The Agreement outlines the complete entrance criteria
required to begin testing.
The Customer soliciting the test verification should ship the product to the tekVizion
PVS, Inc. lab and may, optionally, have a person present to oversee the testing. The
Customer shall ship all the components, which include hardware, software, third
party components and licenses and documentation needed to setup and configure
the product for testing. Before the testing begins the Customer will be given an
opportunity to check and verify the configuration of their product, via remote access
in certain cases. Once testing has been initiated, the products under test are
considered frozen for SIP Verification testing purposes.
2.3 Exit Criteria
To obtain a Statement of Verification and be deemed tekVizion Verified, the devices

under test should have zero defects detected excluding any tests that are not
applicable to the two particular products being tested. Every changed configuration
or software update may necessitate execution of the complete test document to
ensure that no other functionality is compromised. Additional costs may incur If a
new round of testing is necessary to complete testing.
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The following procedures are followed when testing fails:
Preliminary analysis is made to determine the source of the problem. If the

problem is related to the Cisco Unified Communications Manager product not
performing as intended by Cisco, then the problem is reported to Cisco. The
products are still deemed non interoperable and therefore the testing will not
completely pass until this problem is resolved.
If the problem is detected to be in the Customers device under test, the problem
will be logged and reported to the Customer along with call traces captured using
a packet sniffer.
If testing can continue past this failure, the other test cases will be tested and
verified for pass or fail. If the testing cannot progress past this problem, testing
will be halted and a failure report submitted to vendor. The Customer can request
a conference call to discuss the reported problems.
All problems and resolutions encountered during testing are documented in the
final test report.
If the Customers device under test fails to pass all the test cases, then the
product is considered non-interoperable and the vendor will have to apply for a
additional rounds of SIP Verification testing.
The Customer will still receive a final Test Report from tekVizion Labs and
tekVizion Labs will allow for one meeting to discuss the results
The testing is deemed complete under these conditions
All applicable tests in the tekVizion Labs Test Plan have been executed and
successfully passed or,
All applicable tests in the tekVizion Labs Test Plan have been executed and
some of the tests failed.
Any deviations of the test execution or problem acceptance are documented in the
test report.
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Once the device under test successfully passes all the test cases excluding any that
are not applicable, a tekVizion Labs Test Report will be issued along with a
Statement of Verification. The report will contain the products that were used in the
interoperability test, their software/firmware version and the specific test document
along with its version used to perform the testing. The report and the Statement of
Verification will be issued exclusively to the Customer who has contracted the testing
with tekVizion Labs. Its the responsibility of that party to distribute the report as it
sees fit.
You must pass all features that you select to be tested to be considered SIP Verified
and be listed on both the Cisco and tekVizion Web Sites.
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3 Product Overview
Ascom IP-DECT combines the proven DECT standard with VoIP, allowing you to
take advantage of both packet data and high-quality voice connections on the
same network. You can look forward to superb quality of service and excellent
messaging capabilities in a secure radio environment. And because theres
seamless handover, you wont have to worry about interruptions and lost
information, which inevitably result in confusion, misunderstanding and irritation.
IP-DECT offers interference free, dedicated and protected frequency, making it

exceedingly difficult to hack into. Which means you can look forward to minding your
own business safe in the knowledge that mission sensitive information stays where
it belongs.
The Ascom IP-DECT System provides you with a wide range of voice, messaging
and personal alarm handsets purpose built and ideal for office, healthcare and
industry segments.
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4 Executive Summary
The device performed as expected. No critical failures were noted, and SIP Verification
is therefore achieved as a result of this testing activity.
The following summarizes tekVizions findings:
Test Failures
No test failures were noted.
Features not supported
Music on Hold (MOH):

This feature is not supported by the device. For Ascom-to-Ascom calls, the
held party hears a slow-cadence dial tone. For Ascom-to-Cisco calls, the non-
Ascom held party hears silence. For Ascom-to-Cisco calls, if the hold is
initiated by the Cisco phone, the held (Ascom) party hears music.
Multiple Lines per Phone

This feature is not supported by the device.
Call Forking (Bridged Line Appearance)

Communications Manager does not support this feature for 3rd-party devices.
Authentication on registration no password

The device does not support authentication without a password.
Authentication on origination
The device does not support authentication on origination.
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5 Summary of Testing Results
5.1 Features Tested for Interoperability to the Cisco Unified

Communications Manager
Unified Communications RFC Reference To Be Tested?

Manager Feature
Call Hold and Resume 3261, 3264, 2327, 1889 Yes
Transfer Unattended 3261, 3264, 2327, 1889, 3515, 3420, Yes

3265, 3892
Transfer Attended 3261, 3264, 2327, 1889, 3515, 3420, Yes

3265, 3892, 3891
Call Forwarding All 3261, 3264, 2327, 1889 Yes
Call Forwarding No Answer 3261, 3264, 2327, 1889 Yes
Call Forwarding Busy 3261, 3264, 2327, 1889 Yes
Multiple Calls per Line 3261, 3264, 2327, 1889 Yes
Incoming Call Screening 3261, 3264, 2327, 1889, 3725 Yes
Outgoing Call Screening 3261, 3264, 2327, 1889, 3725 Yes
Calling and Connected Line ID 3261, 3264, 2327, 1889, RemotePartyID Yes
Calling and Connected Name ID 3261, 3264, 2327, 1889, RemotePartyID Yes
Message Waiting Indication 3261, 3264, 2327, 1889, 3842 Yes
Three-Way Conference Calling 3261, 3264, 2327, 1889 Yes
Call Forking 3261, 3264, 2327, 1889 Yes
Speed Dialing 3261, 3264, 2327, 1889 Yes
Multiple Lines per Phone 3261, 3264, 2327, 1889 Yes
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5.2 Interpretation of Test Results

The results of this summary are taken from the Test Cases section. The purpose of
this is for CDTP Partner Program Team to review without the need of going into the
full detail of the report. Only those test cases that did not pass are listed in this
section. The full test case listing, including all passed test cases, can be found in
Section 7, Test Cases.
Table 1 Test Results Legend
Result Description
Pass The test case passed with no exceptions
Fail The test case failed details of the failure are noted in the Comments column
N/A The test case is not applicable to the product under test. Justification must be
provided in the Comments column.
N/S Not supported. While the feature tested by this test case generally would be
considered a standard feature for this product category, this specific product (or
this specific release) does not support the feature.
N/T Not tested. The feature is supported by the product under test, but external factors
(lab configuration, e.g.) prevented execution of the test. Justification must be
provided in the Comments column.
Blocked Other test case failures prevented the execution of this test. Reference to the
corresponding failed test case must be provided in the Comments column.
5.3 Installation, Functional, Negative, and Informational Test

Summary
Test Case Description Pass/Fail Comments
5.4 Performance Test Summary
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Test Case Description Pass/Fail Comments
5.5 Items Not Tested

Features that are specific to the internals of the 3rd party product or any features not
listed will not be tested.
Call Forking
Multi-line phone
Authentication with no password
Authentication on origination
Remote reset / restart
SRST support
5.6 Assumptions
Interoperability of 3rd party products Testing will cover only features in 3rd
party products that result in events to and/or from the Cisco Unified
Communications Manager.
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6 Test Environment
6.1 Administration, Testing and Debugging tools

N/A
6.2 Equipment Requirements
Table 2 identifies all equipment/versions used for in this IVT.
Product Version Type Purpose Units Notes
Cisco Products
Cisco Unified 6.1.1 MCS7835 Publisher and 2 3 Lab

Communications Subscriber nodes Provided
Manager
Cisco 5300 (PSTN 12.2(13)T PSTN gateway 1 Lab

GW) Provided
Cisco SIP Phones App: P0S3- 7960 Endpoint 2 Lab

08-2.00 Provided
Boot:
PC030301
DSP:
PS03AT47
Cisco SCCP App: 7960 Endpoint 1 Lab

Phones P00308000 Provided
100
Boot:
PC03A300
Version:
8.0(1.0)
DSP:
4.0(1.0)[A0
]
Unity VoiceMail 4.1 Build Compaq Voice Mail Server 30 ports Lab
4.1(1) Proliant Provided
DL-380
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Product Version Type Purpose Units Notes
rd
3 Party Products
Ascom IP-DECT SW Base 1 Partner

Base Station IPBS[2.3.5] Station Provided
,
Bootcode[4
15],
HW[IPBS1-
A3/2A]
Ascom 9d24 DECT (9d24MKII DECT Device under test 3 handsets, 3 charging Partner
Handset Talker US) Handset bases Provided
SW[3.26],
HW[RAID2
-AAAAD]
Test Tools
Wireshark 0.99.4 Software IP Packet Capture 1 Lab

Provided
Compaq Presario Win XP- Packet captures, 1 Lab

V4000 Pro Laptop Communications Provided
Manager
administrative
access, reporting
Table 2 Equipment and Product Information
6.3 3rd Party Supporting Documentation and Urls

rd
Table 3 3 Party Supporting Documentation
Description Related url
Website
Technical Documentation
User Guide
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Description Related url
Administrator Guide
IVT Questionnaire
Application Note N/A
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6.4 Test Environment Architecture

Figure 1 Test Environment
UCM 6.1
Publisher Subscriber 2
Subscriber 1
Test Tools and

3rd party Cisco IP
Monitoring
endpoints phones
POTS
phones
PSTN
Gateway
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7 Test Cases
This section details the tests that will be performed during the testing period.
Refer to Table 1 Test Results Legend for the definitions of the annotations in the
Pass/Fail column.
DUT Device Under Test

CSP Cisco Skinny Phone
CSIPP Cisco SIP Phone
7.1 Basic Call Scenarios

The intention of this section is to verify that basic calls can be properly handled
between the SIP Phone under test and Cisco Unified Communications Manager.
This test includes the validation of the different call stages from setup, alerting,
connecting, and tear down, as well as different call scenarios between end points, IP
server local or remote extensions and calls to and from PSTN, Cisco SIP and SCCP
phones.
.
Test Description Expected Result Pass/ Comments
Case Fail
Station to Station Calls
7.1.1 DUT to DUT, Two-way voice path, P

originator releases call released properly
call
7.1.2 DUT to DUT, Terminator stops P

originator abandons ringing, originator
call released properly
7.1.3 DUT to DUT, Two-way voice path, P

terminator releases call released properly
7.1.4 DUT to DUT, Busy tone heard at P Call Waiting Disabled

terminator busy originator
7.1.5 DUT to DUT, Ringing at terminator, P

unanswered call ringback at originator,
originator released
properly
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Case Fail
7.1.6 DUT, call to unknown Treatment heard at P

number originator, originator
released properly
7.1.7 DUT to CSP, Two-way voice path, P

call
7.1.8 DUT to CSP, Two-way voice path, P

call
7.1.9 CSP to DUT, Terminator stops P

7.1.10 CSP to DUT, Two-way voice path, P

7.1.11 DUT to CSP, Busy tone heard at P Call Waiting Disabled

7.1.12 DUT to CSP, Ringing at terminator, P

originator released
properly
7.1.13 DUT to CSIPP, Two-way voice path, P

call
7.1.14 DUT to CSIPP, Two-way voice path, P

call
7.1.15 CSIPP to DUT, Terminator stops P

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Case Fail
7.1.16 CSIPP to DUT, Two-way voice path, P

7.1.17 DUT to CSIPP, Busy tone heard at P Call Waiting Disabled

7.1.18 DUT to CSIPP, Ringing at terminator, P

originator released
properly
PSTN Calls
7.1.19 DUT to PSTN, Two-way voice path, P

call
7.1.20 DUT to PSTN, Terminator stops P

7.1.21 DUT to PSTN, Two-way voice path, P

7.1.22 DUT to PSTN, Busy tone heard at P

7.1.23 DUT to PSTN, Ringing at terminator, P

originator released
properly
7.1.24 DUT to PSTN, call to Treatment heard at P

unknown number originator, originator
released properly
7.1.25 PSTN to DUT, PSTN Terminator stops P

abandons call ringing, originator
released properly
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Case Fail
7.1.26 PSTN to DUT, Two-way voice path, P

call
7.1.27 PSTN to DUT, Busy tone heard at P Call Waiting Disabled

7.1.28 PSTN to DUT, Ringing at terminator, P

originator released
properly
DTMF Using G.711 (in band)
7.1.29 DUT retrieves a Voicemail retrieve NS

voicemail, DUT successfully
releases call after
sending DTMF tones
7.1.30 DUT retrieves a Voicemail retrieve NS

voicemail, voicemail successfully
releases call after
receiving DTMF
tones
DTMF Using RFC 2833 (out of band)
7.1.31 DUT retrieves a Voicemail retrieve P

voicemail, DUT successfully
releases call after
sending DTMF tones
7.1.32 DUT retrieves a Voicemail retrieve P

voicemail, voicemail successfully
releases call after
receiving DTMF
tones
Page 23 5/7/08
7.2 Cisco Unified Communications Manager Standard SIP Feature

Support
The goal of this section is to verify protocol interactions between the device under
test and the Cisco Unified Communications Manager standards implementation.
Focus is on feature call functionality, call control and other call information support
interworking capabilities of the endpoint under test and the Cisco Unified
Communications Manager version under test.

Case Fail
Call Forward All (CFA)
7.2.1 DUT to DUT2, Call Two-way audio, call P

forwarded to DUT3, released properly
Endpoint releases
call

DUT3 releases call
7.2.3 DUT to DUT2, Call Terminator stops P

forwarded to DUT3, ringing, call released
DUT abandons call properly
7.2.4 DUT to DUT2, Call Originator hears busy P Call Waiting Disabled
forwarded to DUT3, tone, call released
DUT3 is busy properly
7.2.5 PSTN to DUT, Call Two-way audio, call P

PSTN releases call

DUT2 releases call
7.2.7 PSTN to DUT, Call Terminator stops P

PSTN abandons call properly
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Case Fail
7.2.8 PSTN to DUT, Call Originator hears busy P Call Waiting Disabled on
forwarded to DUT2, tone, call released DUT2
7.2.9 DUT to CSP, Call Two-way audio, call P

DUT releases call
7.2.10 DUT to CSIPP, Call Two-way audio, call P

DUT releases call

DUT2 releases call
7.2.12 DUT to CSIPP, Call Terminator stops P

7.2.13 DUT to CSP, Call Originator hears busy P Call Waiting Disabled on
7.2.14 DUT to CSIPP, Call Originator hears busy P Call Waiting Disabled on

forwarded to CSIPP, released properly
PSTN releases call

forwarded to CSP, released properly
CSP releases call

forwarded to CSIPP, ringing, call released
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Case Fail
forwarded to CSP, tone, call released CSP
CSP is busy properly
Call Forward if Busy (CFB)

Endpoint releases
call

DUT3 releases call

7.2.22 DUT to DUT2, Call Originator hears busy P Call Waiting Disabled on

PSTN releases call

DUT2 releases call


forwarded to DUT2,
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Case Fail
DUT releases call released properly

DUT releases call

DUT2 releases call


PSTN releases call

CSP releases call

Call Forward if No Answer (CFNA)
Page 27 5/7/08

Case Fail

Endpoint releases
call

DUT3 releases call

7.2.40 DUT to DUT2, Call Originator hears busy P Call Waiting Disabled on

PSTN releases call

DUT2 releases call


DUT releases call

DUT releases call
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Case Fail

DUT2 releases call


PSTN releases call

CSP releases call

Call Hold and Resume
7.2.55 DUT to DUT2. DUT2 hears MOH, 2- P CCM does not play MOH for
rd
Originator Holds and way audio resumes 3 party devices. Verified
resumes call for codec G.729. In this
case CCM switches the
codec to G723 while
resuming the call. Ascom
successfully responds to
this with G723 and
Page 29 5/7/08

Case Fail
establishes two-way audio

after resume.
7.2.56 DUT to DUT2. DUT hears MOH, 2- P CCM does not play MOH for
rd
Terminator Holds and way audio resumes 3 party devices.
resumes call
nd rd
Originator Holds call way audio on 2 call, 3 party devices.
nd
to answer an 2 call properly
incoming call released, 2-way
audio resumes
nd rd
Terminator Holds way audio on 2 call, 3 party devices.
nd
call to answer an 2 call properly
audio resumes
nd rd
Originator Holds to way audio on 2 call, 3 party devices.
nd
originate a second 2 call properly
call released, 2-way
audio resumes
nd rd
Terminator Holds to way audio on 2 call, 3 party devices.
nd
audio resumes
7.2.61 DUT to DUT2. DUT2 hears MOH, P CCM does not play MOH for
rd
Originator Holds call, DUT2 leg properly 3 party devices.
Terminator releases released, DUT
before retrieve unable to retrieve,
DUT properly
released
7.2.62 DUT to DUT2. DUT hears MOH, P CCM does not play MOH for
rd
Terminator Holds DUT leg properly 3 party devices.
call, Originator released, DUT2
Page 30 5/7/08

Case Fail
releases before unable to retrieve,

retrieve DUT2 properly
released
7.2.63 PSTN to DUT, PSTN hears MOH, 2- P CCM does not play MOH for
rd
resumes call
7.2.64 PSTN to DUT, PSTN DUT hears MOH, 2- P CCM does not play MOH for
rd
Holds and resumes way audio resumes 3 party devices.
call
nd rd
terminator Holds call way audio on 2 call, 3 party devices.
nd
audio resumes
nd rd
Holds call to answer way audio on 2 call, 3 party devices.
nd
an incoming call 2 call properly
released, 2-way
audio resumes
nd rd
terminator Holds to way audio on 2 call, 3 party devices.
nd
audio resumes
nd rd
Holds to originate a way audio on 2 call, 3 party devices.
nd
second call 2 call properly
released, 2-way
audio resumes
7.2.69 PSTN to DUT, PSTN hears MOH, P CCM does not play MOH for
rd
terminator Holds call, PSTN leg properly 3 party devices.
PSTN releases released, DUT
DUT properly
Page 31 5/7/08

Case Fail
released
7.2.70 PSTN to DUT, PSTN DUT hears MOH, P CCM does not play MOH for
rd
Holds call, End Point DUT leg properly 3 party devices.
release before released, PSTN
retrieve unable to retrieve,
PSTN properly
released
7.2.71 DUT to PSTN, DUT hears MOH, 2- P CCM does not play MOH for
rd
terminator Holds and way audio resumes 3 party devices.
resumes call
7.2.72 DUT to PSTN, PSTN DUT hears MOH, 2- P CCM does not play MOH for
rd
Holds and resumes way audio resumes 3 party devices.
call
nd rd
terminator Holds call way audio on 2 call, 3 party devices.
nd
audio resumes
nd rd
Holds call to answer way audio on 2 call, 3 party devices.
nd
an incoming call 2 call properly
released, 2-way
audio resumes
nd rd
terminator Holds to way audio on 2 call, 3 party devices.
nd
audio resumes
nd rd
Holds to originate a way audio on 2 call, 3 party devices.
nd
second call 2 call properly
released, 2-way
audio resumes
Page 32 5/7/08

Case Fail
7.2.77 DUT to PSTN, DUT PSTN hears MOH, P CCM does not play MOH for
rd
Holds call, PSTN PSTN leg properly 3 party devices.
releases before released, DUT
DUT properly
released
7.2.78 DUT to PSTN, PSTN DUT hears MOH, P CCM does not play MOH for
rd
Holds call, End Point DUT leg properly 3 party devices.
release before released, PSTN
PSTN properly
released
7.2.79 DUT to CSP. CSP hears MOH, 2- P CCM does not play MOH for
rd
Originator Holds and way audio resumes 3 party devices.
resumes call
7.2.80 DUT to CSP. DUT hears MOH, 2- Receives MOH since CCM
Terminator Holds and way audio resumes place the call on hold.
resumes call
nd rd
Originator Holds call way audio on 2 call, 3 party devices.
nd
audio resumes
7.2.82 DUT to CSP. DUT hears MOH, 2- P Receives MOH since CCM
nd
Terminator Holds way audio on 2 call, place the call on hold.
nd
audio resumes
nd rd
Originator Holds to way audio on 2 call, 3 party devices.
nd
audio resumes
Page 33 5/7/08

Case Fail
7.2.84 DUT to CSP. DUT hears MOH, 2- P CCM does not play MOH for
nd rd
nd
audio resumes
7.2.85 DUT to CSP. CSP hears MOH, P CCM does not play MOH for
rd
Originator Holds call, CSP leg properly 3 party devices.
DUT properly
released
7.2.86 DUT to CSP. DUT hears MOH, P CCM does not play MOH for
rd
call, Originator released, CSP
retrieve CSP properly
released
7.2.87 DUT to CSIPP. CSIPP hears MOH, P CCM does not play MOH for
rd
Originator Holds and 2-way audio resumes 3 party devices.
resumes call
7.2.88 DUT to CSIPP. DUT hears MOH, 2- P CCM does not play MOH for
rd
resumes call
P
7.2.89 DUT to CSIPP. CSIPP hears MOH, CCM does not play MOH for
nd rd
Originator Holds call 2-way audio on 2 3 party devices.
nd
to answer an call, 2 call properly
audio resumes
P
7.2.90 DUT to CSIPP. DUT hears MOH, 2- CCM does not play MOH for
nd rd
Terminator Holds way audio on 2 call, 3 party devices.
nd
audio resumes
Page 34 5/7/08

Case Fail
P
nd rd
Originator Holds to 2-way audio on 2 3 party devices.
nd
originate a second call, 2 call properly
audio resumes
P
7.2.92 DUT to CSIPP. DUT hears MOH, 2- CCM does not play MOH for
nd rd
nd
audio resumes
P
rd
Originator Holds call, CSIPP leg properly 3 party devices.
DUT properly
released
P
7.2.94 DUT to CSIPP. DUT hears MOH, CCM does not play MOH for
rd
call, Originator released, CSIPP
retrieve CSIPP properly
released
Call Waiting
7.2.95 DUT to DUT2, Call DUT indicates P

waiting on Originator. incoming call
7.2.96 DUT to DUT2, Call DUT2 indicates P

waiting on incoming call
Terminator.
7.2.97 DUT to PSTN, Call DUT indicates P

Waiting on Originator. incoming call
7.2.98 PSTN to DUT, Call DUT indicates P

Waiting on incoming call
Page 35 5/7/08

Case Fail
Terminator.
7.2.99 DUT to CSP, Call DUT indicates P

7.2.100 CSP to DUT, Call DUT indicates P

Terminator.
7.2.101 DUT to CSIPP, Call DUT indicates P

7.2.102 CSIPP to DUT, Call DUT indicates P

Terminator.
Blind Call Transfer
7.2.103 DUT to DUT2, DUT properly P

Originator transfer to released, 2-way
a second extension audio between DUT2
and extension

Originator failed to released, DUT2
transfer call to a receives treatment,
second extension DUT2 properly
released
7.2.105 DUT to DUT2, All legs properly P

Originator transfer to released
a second extension,
release before
answer
7.2.106 DUT to DUT2, DUT2 properly P

Terminator transfer to released, 2-way
a second extension audio between DUT
and extension
Page 36 5/7/08

Case Fail
Terminator failed to released, DUT

second extension DUT properly
released

Terminator transfer to released
a second extension,
Originator release
before answer

PSTN audio between DUT2
and PSTN

Originator failed to released, DUT2
transfer call to PSTN receives treatment,
DUT2 properly
released

PSTN, release before
answer

PSTN audio between DUT
and PSTN

DUT properly
released

PSTN, Originator
Page 37 5/7/08

Case Fail
release before
answer
7.2.115 PSTN to DUT, DUT properly P

terminator transfer to released, 2-way
second extension audio between PSTN
and extension
7.2.116 PSTN to DUT, DUT properly P

terminator failed to released, PSTN
transfer call receives treatment,
PSTN properly
released
7.2.117 PSTN to DUT, All legs properly P

terminator transfer, released
PSTN releases
before call is
answered
7.2.118 DUT to PSTN, DUT properly P

originator transfer to released, 2-way
second extension audio between PSTN
and extension
7.2.119 DUT to PSTN, DUT properly P

originator failed to released, PSTN
transfer call receives treatment,
PSTN properly
released
7.2.120 DUT to PSTN, All legs properly P

originator transfer, released
PSTN releases
before call is
answered
7.2.121 DUT to CSP, DUT properly P

a second extension audio between CSP
Page 38 5/7/08

Case Fail
and extension

Originator failed to released, CSP
second extension CSP properly
released
7.2.123 DUT to CSP, All legs properly P

a second extension,
release before
answer
7.2.124 DUT to CSP, CSP properly P

and extension

released

a second extension,
Originator release
before answer

PSTN audio between CSP
and PSTN

Originator failed to released, CSP
CSP properly
Page 39 5/7/08

Case Fail
released

answer

and PSTN

DUT properly
released

PSTN, Originator
release before
answer
7.2.133 DUT to CSIPP, DUT properly P

a second extension audio between
CSIPP and extension

Originator failed to released, CSIPP
second extension CSIPP properly
released
7.2.135 DUT to CSIPP, All legs properly P

a second extension,
release before
answer
Page 40 5/7/08

Case Fail
7.2.136 DUT to CSIPP, CSIPP properly P

and extension

released

a second extension,
Originator release
before answer

PSTN audio between
CSIPP and PSTN

Originator failed to released, CSIPP
CSIPP properly
released

answer

and PSTN

Page 41 5/7/08

Case Fail

DUT properly
released

PSTN, Originator
release before
answer
Consultative Call Transfer
7.2.145 DUT to DUT2, DUT2 receives MOH, P

Originator transfer to 2-way audio DUT to
a second extension extension, DUT
properly released, 2-
way audio DUT2 to
extension

Originator failed to DUT receives
transfer call to a treatment, call
second extension, successfully retrieved
retrieve call with 2-way audio,
failed leg properly
released

a second extension, extension, all call
release before legs properly
answer released
7.2.148 DUT to DUT2, DUT receives MOH, P

Terminator transfer to 2-way audio DUT2 to
a second extension extension, DUT2
way audio DUT to
extension
Page 42 5/7/08

Case Fail

Terminator failed to DUT2 receives
failed leg properly
released

Originator release legs properly
before answer released

PSTN PSTN, DUT properly
released, 2-way
audio DUT2 to PSTN

transfer call to PSTN, treatment, call
retrieve call successfully retrieved
with 2-way audio,
failed leg properly
released

PSTN, release before PSTN, all call legs
answer properly released

PSTN PSTN, DUT2
way audio DUT to
PSTN
Page 43 5/7/08

Case Fail

with 2-way audio,
failed leg properly
released

PSTN, Originator PSTN, all call legs
release before properly released
answer
7.2.157 PSTN to DUT, PSTN receives MOH, P

terminator transfer to 2-way audio DUT to
second extension extension, DUT
way audio PSTN to
extension

terminator failed to DUT receives
transfer call treatment, call
successfully retrieved
with 2-way audio,
failed leg properly
released

terminator transfer, 2-way audio DUT to
PSTN releases extension, all call
before call is legs properly
answered released
7.2.160 DUT to PSTN, PSTN receives MOH, P

originator transfer to 2-way audio DUT to
second extension extension, DUT
way audio PSTN to
Page 44 5/7/08

Case Fail
extension

originator failed to DUT receives
transfer call treatment, call
with 2-way audio,
failed leg properly
released

originator transfer, 2-way audio DUT to
PSTN releases extension, all call
before call is legs properly
answered released
7.2.163 DUT to CSP, CSP receives MOH, P

way audio CSP to
extension

failed leg properly
released

release before legs properly
answer released

Terminator transfer to 2-way audio DUT to
Page 45 5/7/08

Case Fail
way audio CSP to

extension

Terminator failed to DUT receives
failed leg properly
released

Terminator transfer to 2-way audio DUT to
Originator release legs properly

PSTN PSTN, DUT properly
released, 2-way
audio CSP to PSTN

with 2-way audio,
failed leg properly
released

PSTN, release before PSTN, all call legs
answer properly released
7.2.172 DUT to CSP, DUT receives MOH, P

Terminator transfer to 2-way audio CSP to
PSTN PSTN, CSP properly
released, 2-way
Page 46 5/7/08

Case Fail
audio DUT to PSTN

Terminator failed to CSP receives
with 2-way audio,
failed leg properly
released

Terminator transfer to 2-way audio CSP to
PSTN, Originator PSTN, all call legs
answer
7.2.175 DUT to CSIPP, CSIPP receives P

Originator transfer to MOH, 2-way audio
a second extension DUT to extension,
DUT properly
released, 2-way
audio CSIPP to
extension

Originator failed to MOH, DUT receives
failed leg properly
released

a second extension, DUT to extension, all
release before call legs properly
answer released

Terminator transfer to MOH, 2-way audio
DUT to extension,
Page 47 5/7/08

Case Fail
a second extension DUT properly

released, 2-way
audio CSIPP to
extension

Terminator failed to MOH, DUT receives
failed leg properly
released

Terminator transfer to MOH, 2-way audio
a second extension, DUT to extension, all
Originator release call legs properly

PSTN DUT to PSTN, DUT
way audio CSIPP to
PSTN

with 2-way audio,
failed leg properly
released

PSTN, release before DUT to PSTN, all call
answer legs properly
released
Page 48 5/7/08

Case Fail
7.2.184 DUT to CSIPP, DUT receives MOH, P

Terminator transfer to 2-way audio CSIPP
PSTN to PSTN, CSIPP
way audio DUT to
PSTN

Terminator failed to CSIPP receives
with 2-way audio,
failed leg properly
released

Terminator transfer to 2-way audio CSIPP
PSTN, Originator to PSTN, all call legs
answer
3-Way Conference
NS
7.2.187 DUT to DUT2, DUT2 receives MOH,
Originator is the DUT receives
bridge ringback, 3-way
audio
NS
bridged call to a treatment, call
DUT3, retrieve call successfully retrieved
with 2-way audio
NS
bridge, Terminator ringback, DUT2
release before properly released, 2-
answer way audio DUT
DUT3 after answer
Page 49 5/7/08

Case Fail
NS
bridge, Originator ringback, call
cancel 3 way call, successfully retrieved
retrieve original call with 2-way audio
7.2.191 DUT to DUT2, DUT receives MOH, NS

Terminator is the DUT2 receives
audio
NS
7.2.192 DUT to DUT2, DUT receives MOH,
bridged call, retrieve treatment, call
original call successfully retrieved
with 2-way audio
NS
Terminator bridged to DUT2 receives
a DUT3, Originator ringback, DUT
releases before properly released, 2-
answer way audio DUT2
DUT3 after answer
NS
Originator bridge to DUT receives
PSTN ringback, 3-way
audio
NS
bridge call to PSTN, treatment, call
with 2-way audio
NS
PSTN, terminator ringback, DUT2
Page 50 5/7/08

Case Fail
PSTN after answer

NS
Terminator bridge to DUT2 receives
audio
NS
with 2-way audio
NS
Terminator bridge to DUT2 receives
PSTN, Originator ringback, DUT
answer way audio DUT2
PSTN after answer
NS
7.2.200 PSTN to DUT, PSTN receives MOH,
terminator bridge to DUT receives
DUT2 ringback, 3-way
audio
NS
terminator failed to DUT receives
bridge call treatment, call
with 2-way audio
NS
terminator attempt to DUT receives
bridge, PSTN ringback, PSTN
releases before call is properly released, 2-
answered way audio DUT
DUT2 after answer
NS
7.2.203 DUT to PSTN, PSTN receives MOH,
originator bridge to DUT receives
ringback, 3-way
Page 51 5/7/08

Case Fail
DUT2 audio
7.2.204 DUT to PSTN, PSTN receives MOH, NS

originator failed to DUT receives
bridge call treatment, call
with 2-way audio
7.2.205 DUT to PSTN, PSTN receives MOH, NS

originator attempts to DUT receives
bridge call, PSTN ringback, PSTN
releases before call is properly released, 2-
answered way audio DUT
DUT2 after answer
NS
7.2.206 DUT to CSP, CSP receives MOH,
audio
NS
with 2-way audio
NS
bridge, Terminator ringback, CSP
DUT2 after answer
7.2.209 DUT to CSP, CSP receives MOH, NS


Terminator is the CSP receives
Page 52 5/7/08

Case Fail

audio

with 2-way audio

Terminator bridged to CSP receives
answer way audio CSP
DUT2 after answer
7.2.213 DUT to CSP, CSP receives MOH, NS

audio
NS
with 2-way audio
NS
PSTN, terminator ringback, CSP
PSTN after answer
NS
7.2.216 DUT to CSP, DUT receives MOH,
Terminator bridge to CSP receives
audio

Page 53 5/7/08

Case Fail

with 2-way audio

Terminator bridge to CSP receives
answer way audio CSP
PSTN after answer
NS
7.2.219 DUT to CSIPP, CSIPP receives
Originator is the MOH, DUT receives
audio
NS
with 2-way audio
NS
bridge, Terminator ringback, CSIPP
DUT2 after answer
NS

Terminator is the CSIPP receives
audio
Page 54 5/7/08

Case Fail

with 2-way audio

Terminator bridged to CSIPP receives
answer way audio CSIPP
DUT2 after answer
7.2.226 DUT to CSIPP, CSIPP receives NS

Originator bridge to MOH, DUT receives
audio

with 2-way audio

Originator bridge to MOH, DUT receives
PSTN, terminator ringback, CSIPP
PSTN after answer

Terminator bridge to CSIPP receives
audio

Page 55 5/7/08

Case Fail
with 2-way audio

Terminator bridge to CSIPP receives
answer way audio CSIPP
PSTN after answer
Calling Line Identification
7.2.232 DUT to DUT2, Calling Calling Line ID P

Line identification presented at
type I terminator
7.2.233 DUT to CSP, Calling Calling Line ID P

type I terminator
7.2.234 DUT to CSIPP, Calling Line ID P

Calling Line presented at
identification type I terminator
7.2.235 CSIPP to DUT2, Calling Line ID P

7.2.236 CSP to DUT2, Calling Calling Line ID P

type I terminator
7.2.237 DUT to PSTN, Calling Line ID P

7.2.238 PSTN to DUT, Calling Line ID P

7.2.239 DUT to DUT2, Type II Calling Line ID P

on terminator, call in presented at
Page 56 5/7/08

Case Fail
progress between terminator

two end points

on originator, call in presented at
two end points

on originator, call in presented at
End point and PSTN

on terminator, call in presented at
PSTN and End point
7.2.243 DUT to DUT2, Calling Calling Line ID not N/A

type I Restricted terminator
7.2.244 DUT to CSP, Calling Calling Line ID not P

7.2.245 DUT to CSIPP, Calling Line ID not P

Restricted
7.2.246 CSIPP to DUT2, Calling Line ID not P

Restricted
7.2.247 CSP to DUT2, Calling Calling Line ID not P

7.2.248 Multiline DUT (single Calling Line ID NS
Page 57 5/7/08

Case Fail
line presented, others presented at

not) to DUT Line terminator
Presented (Calling
party)
7.2.249 Multiline DUT (single Calling Name NS

not) DUT to DUT terminator
Name Presented
(Calling Party)
7.2.250 Multiline DUT (single Calling Line ID and NS

line presented, others Calling Name
not) DUT to DUT line presented at
and Name Presented terminator
(Calling Party)
7.2.251 Multiline DUT (single Calling Line ID not NS

not) DUT to DUT Line terminator
restricted (Calling
Party)
7.2.252 Multiline DUT (single Calling Name not NS

Name restricted
(Calling Party)
7.2.253 Multiline DUT (single Calling Line ID and NS

line presented, others Name not presented
not) DUT to DUT Line at terminator
and Name restricted
(Calling Party)
7.2.254 Call Forward (DUT to Calling Line ID P

DUT2, forwarded to presented at DUT3
DUT3)
7.2.255 Call Transfer (DUT to Calling Line ID P

DUT2, originator
Page 58 5/7/08

Case Fail
transfer to DUT3) presented at DUT3
7.2.256 Call Transfer (DUT to Calling Line ID P

DUT2, terminator presented at DUT3
transfer to DUT3)
Calling Name Presentation
7.2.257 DUT to DUT2, Calling Calling Name CCM does not support
Party Name presented at calling name presentation
rd
terminator NT on 3 party devices.
7.2.258 DUT to DUT2, Calling Calling Name not CCM does not support
Party Name presented at calling name presentation
rd
Restricted terminator NT on 3 party devices.
7.2.259 DUT to CSP, Calling Calling Name P

Party Name presented at
terminator
7.2.260 DUT to CSP, Calling Calling Name not P

Restricted terminator
7.2.261 DUT to CSIPP, Calling Name P

Calling Party Name presented at
terminator
7.2.262 DUT to CSIPP, Calling Name not P

7.2.263 CSP to DUT, Calling Calling Name P

terminator
7.2.264 CSP to DUT, Calling Calling Name not P

7.2.265 CSIPP to DUT, Calling Name P
Page 59 5/7/08

Case Fail

terminator
7.2.266 CSIPP to DUT, Calling Name not P

7.2.267 DUT to PSTN, Calling Name NT

terminator PSTN does not support this
7.2.268 PSTN to DUT, Calling Name NT

terminator PSTN does not support this
7.2.269 Multline DUT (single Calling Line ID NS

not) to DUT Line terminator
Presented (Calling
party)
7.2.270 Multline DUT (single Calling Name NS

Name Presented
(Calling Party)
7.2.271 Multline DUT (single Calling Line ID and NS

line presented, others Calling Name
not) DUT to DUT line presented at
and Name Presented terminator
(Calling Party)
7.2.272 Multline DUT (single Calling Line ID not NS

not) DUT to DUT Line terminator
restricted (Calling
Party)
7.2.273 Multline DUT (single Calling Name not NS

Page 60 5/7/08

Case Fail

Name restricted
(Calling Party)
7.2.274 Multline DUT (single Calling Line ID and NS

line presented, others Calling Name not
not) DUT to DUT Line presented at
and Name restricted terminator
(Calling Party)
7.2.275 Call Forward (DUT to Calling Name P

DUT2, forwarded to presented at DUT3
DUT3)
7.2.276 Call Transfer (DUT to Calling Name P

DUT2, originator presented at DUT3
transfer to DUT3)
7.2.277 Call Transfer (DUT to Calling Name P

DUT2, terminator presented at DUT3
transfer to DUT3)
Multiple Lines per Phone
7.2.278 DUT1 line 1 calls 2-way audio on each NS

DUT2 line 1, DUT1 call. Held calls
line 2 calls DUT2 line receive MOH.
2, etc. (all DUT1 lines Connected Line ID
occupied). Alternate and Name display
between calls. reflects current call.
7.2.279 DUT line 1 calls CSP, 2-way audio on each NS

DUT line 2 calls CSP, call. Held calls
etc. (all DUT1 lines receive MOH.
occupied). Alternate Connected Line ID
between calls. and Name display
reflects current call.
7.2.280 DUT line 1 calls 2-way audio on each NS

CSIPP, DUT line 2 call. Held calls
calls CSIPP, etc. (all receive MOH.
Page 61 5/7/08

Case Fail
DUT1 lines Connected Line ID

occupied). Alternate and Name display
between calls. reflects current call.
7.2.281 DUT2 calls DUT1 line 2-way audio on each NS

1, CSP calls DUT1 call. Held calls
line 2, CSIPP calls receive MOH. Calling
DUT1 line 3, PSTN Line ID and Name
calls DUT1 line 4. display reflects
Alternate between current call. All calls
calls. All originators properly released.
release.
7.2.282 DUT1 line 1 calls 2-way audio on each NS

DUT2, DUT1 line 2 call. Held calls
calls CSP, DUT1 line receive MOH.
3 calls CSIPP, DUT1 Connected Line ID
line 4 calls PSTN. and Name display
Alternate between reflects current call.
calls. All originators All calls properly
release. released.
7.2.283 Line Busy: CFWD 2-way audio between NS

Busy Line 1 to Line 2, CSP and Line 3.
CFWD Busy Line 2 to
Line 3, Line 1 in call
with Line 2, CSP calls
Line 1
7.2.284 CSP calls DUT line 1. 3-way audio between NS

CSIPP calls DUT line parties.
2. Place line 1 on
hold, answer line 2.
Bring line 1 into
conference.
7.2.285 CSP calls DUT line 1. Line 1 properly NS

Blind transfer to line released. 2-way
2. Pick up line 2. audio with Line 2.
Page 62 5/7/08

Case Fail
7.2.286 MWI Single line: Line 2 message NS

leave VMAIL on line waiting is indicated,
2, retrieve and delete indicator cleared
line 2 VMAIL when VMAIL deleted
7.2.287 MWI All lines: leave MWI consistent with NS

VMAIL on all lines, voice mail state per
retrieve and delete line.
each
Call Forking
7.2.288 DUT1 to DUT2, 2-way audio, DUT3 N/A Call forking is not supported
rd
DUT3 Call Forking ring stop for 3 party devices.
DUT2 Answers
rd
DUT3 Call Forking ring stop for 3 party devices.
DUT3 Answers
7.2.290 DUT1 to DUT2, 2-way audio, CSIPP N/A Call forking is not supported
rd
CSIPP Call Forking ring stop for 3 party devices.
DUT2 Answers.
rd
CSIPP Call Forking ring stop for 3 party devices.
CSIPP Answers.
7.2.292 DUT1 to DUT2, CSP 2-way audio, CSP N/A Call forking is not supported
rd
Call Forking DUT2 ring stop for 3 party devices.
Answers.
7.2.293 DUT1 to DUT2, CSP 2-way audio, DUT2 N/A Call forking is not supported
rd
Call Forking CSP ring stop for 3 party devices.
Answers.
7.2.294 DUT1 to DUT2, 2-way audio, other N/A Call forking is not supported
rd
DUT3 Call Forking, phone ring stop for 3 party devices.
Race condition
DUT2 and DUT3
Page 63 5/7/08

Case Fail
Answer
7.2.295 DUT1 to DUT2, 2-way audio, other N/A Call forking is not supported
rd
CSIPP Call Forking, phone ring stop for 3 party devices.
Race condition
DUT2 and CSIPP
Answer
7.2.296 DUT1 to DUT2, CSP 2-way audio, other N/A Call forking is not supported
rd
Call Forking, Race phone ring stop for 3 party devices.
condition DUT2 and
CSP Answer
7.2.297 DUT1 to DUT2, Only DUT3 rings, 2- N/A Call forking is not supported
rd
DUT3 Call Forking way audio DUT1 to for 3 party devices.
DUT2 CFWD all to DUT3
DUT3. DUT3
Answers
7.2.298 DUT1 to DUT2, Only DUT3 rings, 2- N/A Call forking is not supported
rd
DUT3 Call Forking way audio DUT1 to for 3 party devices.
DUT2 Busy. DUT3 DUT3
Answers
7.2.299 Call Forking Two lines on DUT N/A Call forking is not supported
rd
Multiline DUT ring, 2-way audio on for 3 party devices.
answered line, ring
stop on other line
7.2.300 Call Forking Call waiting indication N/A Call forking is not supported
rd
Multicall DUT (Call on in-use line, 2-way for 3 party devices.
waiting) audio on answered
line, call waiting
indication ends when
call answered
Message Waiting Indicator
7.2.301 Call DUT, leave voice MWI activated

mail
Page 64 5/7/08

Case Fail
7.2.302 Retrieve only voice MWI cleared

mail
7.2.303 Call DUT, leave 2 MWI activates with

voice mails, retrieve first voice mail, stays
one, hang up, active after first
retrieve second voice retrieve, clears after
mail, hang up second retrieve NS Unity does not support this
7.2.304 Active call, second MWI activates while

inbound call goes to in call
VM, leave voice mail
Speed Dial
7.2.305 Speed dial max digits INVITE sent with all P Max 22 digits supported by
(25) 25 digits device.
7.2.306 Configure and use a INVITE sent with all P

speed dial assigned digits
to a 25 digit number,
with external access
number prefix (9).
7.2.307 Speed dial min digits INVITE sent with all P

(4 digit ext): digits
7.2.308 Configure and use a INVITE sent with all P

speed dial to a 4 digit digits
number
7.2.309 Speed dial no entry Call does not P Device does not permit null
originate speed dial configuration
7.2.310 Max Speed dial All speed dial entries P

entries. Use first and populated and usable
last entries, and one
other
7.2.311 Exceed Max Speed Entry n+1 rejected, all P

other entries remain
Page 65 5/7/08

Case Fail
Dial entries. intact and usable
7.2.312 Re-assign existing Number successfully P

speed dial to new changed and usable
number (25 digit to 4
digit)
7.2.313 Delete Speed Dial Number no longer P

Entry. usable. Remaining
entries unchanged.
7.2.314 Reset phone Speed dial P

configuration persists
through reset
7.2.315 Power cycle phone Speed dial P

configuration persists
through power cycle
7.3 System Control and Verification

These tests are executed to determine the impact on calls, the Communications
Manager and the 3rd party application when combinations of the aforementioned fail
by power failure or network connectivity problems. Testing robustness of the
application through hardware and software fault insertion i.e. failover/failback.

Case Fail
Registration and Digest Authentication (Basic)
7.3.1 DUT Authenticated DUT registers N/A

on Registration successfully
(name only)
(Positive)
7.3.2 DUT Authenticated DUT registration N/A

on Registration rejected, retries
(name Only)
Page 66 5/7/08

Case Fail
(Negative)
7.3.3 DUT Authenticated DUT registers P

on Registration successfully
(name and password)
(Positive)
7.3.4 DUT Authenticated DUT registration P

on Registration rejected, retries
(name and password)
(Negative)
7.3.5 DUT Authenticated DUT resends INVITE N/A

on Origination with Authorization
header, successfully
originates call
7.3.6 DUT Re-Registers Re-registration P

before Registration successful, DUT can
Time Expires originate calls
7.3.7 DUT Responds to DUT maintains N/A

Keep-Alive registration
7.3.8 Restart DUT phone DUT restarts and N/A

remotely registers successfully
7.3.9 DUT Multiline All lines register NS

registration successfully and can
originate calls
7.3.10 Loses network DUT can originate P

connection then re- calls after registration
connected
Page 67 5/7/08
7.4 SIP Endpoint Functionality Verification

These tests are executed to verify specific information about the third-party product
to Cisco. This is in relation to the IVT Questionnaire supplied by the vendor. Test
cases in this section will be selected or modified to reflect attributes of the device
under test. Standard examples are:

Case Fail
Voice Codec Support
7.4.1 G.711 -law P
7.4.2 G.711 A-law P
7.4.3 G.723 NT Not tested per Cisco.
7.4.4 G.729 P
7.4.5 Packetization period NT This cannot be changed in

CCM.
7.4.6 Codec negotiation P
7.4.7 Midcall codec P One G.729 leg, one G.711

renogiation leg conferenced.
General Phone Functions
7.4.8 Phone display P

(missed calls, called
numbers, received
calls)
7.4.9 All visible buttons and P

soft keys function as
labelled
7.4.10 User configuration P

options
Page 68 5/7/08

Case Fail
General Dial Services
7.4.11 Redial simple NS
7.4.12 Last Call Return NS
7.5 Support for SIP-based RFCs and Drafts

The goal of this section is to verify specific features and functions based on SIP
related RFCs and draft specifications. These tests only apply to those features that
are supported.
All calls should stay connected for at least the duration of the session timer.

Case Fail
Diversion Header Indication Support
7.5.1 ISDN to SIP, GW NT

sends INVITE with
Diversion info, CFU

sends INVITE with
Diversion info,
CFTOD

sends INVITE with
Diversion info, CFB

sends INVITE with
Diversion info, CFNA

sends INVITE with
Diversion info,
Page 69 5/7/08

Case Fail
CFUNV
7.5.6 SIP to ISDN, NT

CallManger sends
INVITE with
Diversion info, CFU
7.5.7 SIP to ISDN, NT

CallManger sends
INVITE with
Diversion info,
CFTOD
7.5.8 SIP to ISDN,

CallManger sends
INVITE with CUCM does not send
Diversion info, CFB N/A Diversion header to GW
7.5.9 SIP to ISDN,

CallManger sends
INVITE with CUCM does not send
Diversion info, CFNA N/A Diversion header to GW
7.5.10 SIP to ISDN,

CallManger sends
INVITE with
Diversion info, CUCM does not send
CFUNV N/A Diversion header to GW
Privacy Headers, Remote-Party-ID Headers
7.5.11 PSTN to DUT, GW P

sends INVITE with
Remote-Party-ID
Header
7.5.12 DUT to PSTN, GW P

sends INVITE with
Remote-Party-ID
Header
Page 70 5/7/08
7.6 Reliability and Stability

The vendors product will be tested for general products reliability , stability . and
redundancy capabilities. Tests will verify the ability of the product to support
Communications Manager redundancy, architecture, and stability with repetitive calls
(simultaneously or independent) in a constant manner with various hold times of up to
24 hours without any operational defects or loss of audio quality. The goal for these tests
is to make sure the phone has some mechanism that in case of a "SUB" failure, it can re-
register to the other "SUB", without any manual intervention.

Case Fail
Primary Call Control Failure
7.6.1 DUT Registers with P

secondary server
when the primary is
unavailable
7.6.2 DUT re-Registered P

when primary is back
in service
7.6.3 Multi-line DUT NS

Registers with
secondary server
when the primary is
unavailable
7.6.4 Multi-line DUT re- NS

Registered when
primary is back in
service
7.6.5 Basic call, DUT uses P

secondary server
when primary is
unavailable
7.6.6 Call to PSTN using P

secondary server
when primary is
unavailable
Page 71 5/7/08

Case Fail
7.6.7 Call from PSTN, call P

is terminated by
secondary when
primary is unavailable
7.6.8 Active call, failover P

occurs
7.6.9 Failover occurs P

during call setup
7.6.10 DUT Registers with NT SRST is not available in the

SRST server when lab.
the CCM is
unavailable
7.6.11 DUT re-Registered NT SRST is not available in the

when CCM is back lab.
on service
7.6.12 Basic call, DUT uses NT SRST is not available in the

CCM is unavailable
7.6.13 Call to PSTN using NT SRST is not available in the

CCM is unavailable
7.6.14 Call from PSTN, call NT SRST is not available in the

is terminated by lab.
SRST when CCM is
unavailable
7.6.15 Active call, failover NT SRST is not available in the

occurs lab.
Recovery from Loss of Power
7.6.16 Remove power from Call lost, device P

DUT with active call recovers and can
originate / terminate
Page 72 5/7/08

Case Fail
new calls when

power restored
without test engineer
intervention
Recovery from Loss of Uplink Connectivity
7.6.17 Network Bounce LAN connection P

DUT idle loss/recovery within
10 seconds
7.6.18 Network outage LAN connection P

DUT idle loss/recovery within 3
minutes
7.6.19 Network Bounce LAN connection NS Multiple lines are not

Multi-line DUT loss/recovery within supported on DUT.
10 seconds
7.6.20 Network outage LAN connection NS Multiple lines are not

Multi-line DUT loss/recovery within 3 supported on DUT.
minutes

DUT in-call loss/recovery within
10 seconds

DUT in-call loss/recovery within 3
minutes

DUT with call on-hold loss/recovery within
10 seconds

DUT with call on-hold loss/recovery within 3
minutes

DUT in ring state loss/recovery within
Page 73 5/7/08

Case Fail
10 seconds

DUT in ring state loss/recovery within 3
minutes
7.7 SIP Torture Tests

The tests in this section are taken from IETF RFC 4475. Some of the messages are
valid, some are not. The RFC tends to be vague in prescribing specific required
responses to these messages. In general, however, the following results are
expected:
No message shall cause the device under test to crash, reset, reboot, or exhibit
any other kind of general failure that prevents correct processing of subsequent
messages.
The device shall detect invalid messages as such and shall either provide an
appropriate negative response or simply discard the message.
The device shall not reject valid messages as badly-formed (although the
message may be rejected for other, semantic reasons e.g. the device does not
support REGISTER messages).
Note: These tests are not performed as part of the basic testing. They will be
performed only if the partner request explicitly and are charged separately.

Case Fail
Valid Messages
7.7.1 A short tortuous Message accepted NT

INVITE by DUT
7.7.2 Wide range of valid Message accepted NT

characters by DUT
7.7.3 Valid use of the % Message accepted NT
Page 74 5/7/08

Case Fail
escaping mechanism by DUT
7.7.4 Escaped nulls in Message accepted NT

URIs by DUT
7.7.5 Use of % when it is Message accepted NT

not an escape by DUT
7.7.6 Message with no Message accepted NT

LWS between display by DUT
name and <
7.7.7 Long values in Message accepted NT

header fields by DUT
7.7.8 Extra trailing octets in Message accepted NT

a UDP datagram by DUT
7.7.9 Semicolon separated Message accepted NT

parameters in URI by DUT
user part
7.7.10 Varied and unknown Message accepted NT

transport types by DUT
7.7.11 Multipart MIME Message accepted NT

message by DUT
7.7.12 Unusual reason Message accepted NT

phrase by DUT
7.7.13 Empty reason phrase Message accepted NT

by DUT
Invalid Messages
7.7.14 Extraneous header Message rejected by NT

field separators DUT
7.7.15 Content length larger Message rejected by NT

than message DUT
Page 75 5/7/08

Case Fail
7.7.16 Negative Content- Message rejected by NT

Length DUT
7.7.17 Request scalar fields Message rejected by NT

with overlarge values DUT
7.7.18 Response scalar fields Message rejected by NT

with overlarge values DUT
7.7.19 Unterminated quoted Message rejected by NT

string in display-name DUT
7.7.20 enclosing Request- Message rejected by NT

URI DUT
7.7.21 Malformed SIP Message rejected by NT

Request-URI DUT
(embedded LWS)
7.7.22 Multiple SP separating Message rejected by NT

Request-Line DUT
elements
7.7.23 SP characters at end Message rejected by NT

of Request-Line DUT
7.7.24 Escaped headers in Message rejected by NT

SIP Request-URI DUT
7.7.25 Invalid timezone in Message rejected by NT

Date header field DUT
7.7.26 Failure to enclose Message rejected by NT

name-addr URI in <> DUT
7.7.27 Spaces within addr- Message rejected by NT

spec DUT
7.7.28 Non-token characters Message rejected by NT

in display-name DUT
Page 76 5/7/08

Case Fail
7.7.29 Unknown protocol Message rejected by NT

version DUT
7.7.30 Start line and CSeq Message rejected by NT

method mismatch DUT
7.7.31 Unknown Method with Message rejected by NT

CSeq method DUT
mismatch
7.7.32 Overlarge response Message rejected by NT

code DUT
Transaction Layer Semantics
7.7.33 Missing transaction Message rejected by NT

identifier DUT
Application Layer Semantics
7.7.34 Missing Required NT

Header Fields
7.7.35 Request-URI with NT

unknown scheme
7.7.36 Request-URI with NT

known but atypical
scheme
7.7.37 Unknown URI NT

schemes in header
fields
7.7.38 Proxy-Require and NT

Require
7.7.39 Unknown Content- NT

Type
7.7.40 Unknown NT
Page 77 5/7/08

Case Fail
authorization scheme
7.7.41 Multiple values in NT

single value required
fields
7.7.42 Multiple Content- NT

Length values
7.7.43 200 OK Response NT

with broadcast Via
header field value
7.7.44 Max-Forwards of NT
zero
7.7.45 REGISTER with a NT

contact header
parameter
7.7.46 REGISTER with a url NT

parameter
7.7.47 REGISTER with a url NT

escaped header
7.7.48 Unacceptable Accept NT

offering
Backward Compatibility
7.7.49 INVITE with RFC NT

2543 syntax
Page 78 5/7/08

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tekVizion Confidential

SIP Verification Test Plan and

Revision Author Date Comment

0.1 Gerry Pearson 8/15/2006 Initial draft

0.2 Abelardo Guajardo 9/7/2006 DTMF modifications

0.3 Srikanth Palla 04.24/2008 Updated with test results

1.1 Document Scope

1.2 tekVizion Labs

tekVizion Certification Testing

tekVizion SIP Verification Testing

Third-Party Certification/Validation Testing

PBX/IP PBX Trunk Interface Interoperability Testing

IMS Conformance / Interoperability Testing

1.3 tekVizion PVS, Inc.

tekVizions core services include

tekVizion helps service providers achieve a smooth transition to packet-voice

2 Testing Process and Criteria

2.1 Test Approach

2.2 Entry Criteria

2.3 Exit Criteria

To obtain a Statement of Verification and be deemed tekVizion Verified, the devices

The following procedures are followed when testing fails:

Preliminary analysis is made to determine the source of the problem. If the

The testing is deemed complete under these conditions

IP-DECT offers interference free, dedicated and protected frequency, making it

The following summarizes tekVizions findings:

Features not supported

Music on Hold (MOH):

Multiple Lines per Phone

Call Forking (Bridged Line Appearance)

Authentication on registration no password

5 Summary of Testing Results

5.1 Features Tested for Interoperability to the Cisco Unified

Unified Communications RFC Reference To Be Tested?

Call Hold and Resume 3261, 3264, 2327, 1889 Yes

Transfer Unattended 3261, 3264, 2327, 1889, 3515, 3420, Yes

Transfer Attended 3261, 3264, 2327, 1889, 3515, 3420, Yes

Call Forwarding All 3261, 3264, 2327, 1889 Yes

Call Forwarding No Answer 3261, 3264, 2327, 1889 Yes

Call Forwarding Busy 3261, 3264, 2327, 1889 Yes

Multiple Calls per Line 3261, 3264, 2327, 1889 Yes

Incoming Call Screening 3261, 3264, 2327, 1889, 3725 Yes

Outgoing Call Screening 3261, 3264, 2327, 1889, 3725 Yes

Message Waiting Indication 3261, 3264, 2327, 1889, 3842 Yes

Three-Way Conference Calling 3261, 3264, 2327, 1889 Yes

Call Forking 3261, 3264, 2327, 1889 Yes

Speed Dialing 3261, 3264, 2327, 1889 Yes

Multiple Lines per Phone 3261, 3264, 2327, 1889 Yes

5.2 Interpretation of Test Results

Table 1 Test Results Legend

Pass The test case passed with no exceptions

5.3 Installation, Functional, Negative, and Informational Test

Test Case Description Pass/Fail Comments

5.4 Performance Test Summary

Test Case Description Pass/Fail Comments

5.5 Items Not Tested

6.1 Administration, Testing and Debugging tools

6.2 Equipment Requirements

Table 2 identifies all equipment/versions used for in this IVT.

Product Version Type Purpose Units Notes

Cisco Unified 6.1.1 MCS7835 Publisher and 2 3 Lab

Cisco 5300 (PSTN 12.2(13)T PSTN gateway 1 Lab

Cisco SIP Phones App: P0S3- 7960 Endpoint 2 Lab

Cisco SCCP App: 7960 Endpoint 1 Lab