ALE Application Partner Program Inter-Working Report

Transcription

1 ALE Application Partner Program Inter-Working Report Partner: Cetis Application type: VoIP SIP Hotel Phone Application name: 9600 IP, 3300 IP, E100IP, 3302IPTRM, NDC2110S, M203IP, 9602 IP, E203IP 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 issues a new major release of such Alcatel-Lucent Enterprise product (incorporating new features or functionalities), whichever first occurs. ALE MAKES NO REPRESENTATIONS, WARRANTIES OR CONDITIONS WITH RESPECT TO THE APPLICATION PARTNER PRODUCT. WITHOUT LIMITING THE GENERALITY OF THE FOREGOING, ALE 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 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 1/57

2 Certification overview Date of the certification February 2018 ALE International representative AAPP member representative Alcatel-Lucent Enterprise Communication Platform Alcatel-Lucent Enterprise Communication Platform release AAPP member application release Application Category Rachid Himmi Sid Bray OmniPCX Enterprise R12.1 M d 9602 IP - CD IPTRM - CT IP - CD IP - C E100IP - CC E203IP - CD NDC2110SN - CD M203IP - CD Terminals Hospitality dedicated hardware Author(s): Reviewer(s): Karthik Padmarajan Rachid Himmi, Thierry Chevert, Krassimira Atanassov Revision History Edition 1: creation of the document Janurary Test results Passed Refused Postponed Passed with restrictions Refer to the section 6 for a summary of the test results. IWR validity extension Extension for the all range since they use the same SIP stack: ALE Application Partner Program Inter-working report - Edition 1 - page 2/57

3 AAPP Member Contact Information Contact name: Title: Address: Sid Bray Development and Support 4975 No. 30 th Street Zip Code: City: Colorado Springs Country: USA Phone: Fax: Mobile Phone: Web site: address: ALE Application Partner Program Inter-working report - Edition 1 - page 3/57

4 TABLE OF CONTENTS 1 INTRODUCTION VALIDITY OF THE INTERWORKING REPORT LIMITS OF THE TECHNICAL SUPPORT CASE OF ADDITIONAL THIRD PARTY APPLICATIONS APPLICATION INFORMATION TEST ENVIRONMENT HARDWARE CONFIGURATION SOFTWARE CONFIGURATION SUMMARY OF TEST RESULTS SUMMARY OF MAIN FUNCTIONS SUPPORTED SUMMARY OF PROBLEMS SUMMARY OF LIMITATIONS NOTES, REMARKS TEST RESULT TEMPLATE TEST RESULTS GENERIC TESTS Phone initialization, SIP registration and authentication Audio codecs negotiations Defense Basic call Local telephonic features Other features HOTEL / HOSPITAL TESTS Provisioning Check-in and check-out Do not disturb Forward Voice mail Wake-up Calls Change room state Mini bar Hotel Calls Multi-line Multi-occupation Additional Tests APPENDIX A : AAPP MEMBER S APPLICATION DESCRIPTION APPENDIX B: CONFIGURATION REQUIREMENTS OF THE AAPP MEMBER S APPLICATION VOIP SERIES CONFIGURATION CONFIGURE IP ADDRESS ACCESS THE WEBUI CONFIGURE DNS CONFIGURE NTP CONFIGURE TRANSPORT PROTOCOLS CONFIGURE SIP INTERFACE SETTINGS CONFIGURE SIP REGISTRAR SETTINGS CONFIGURE CODECS CONFIGURE SERVICE SETTINGS CALL FORWARDING ALE Application Partner Program Inter-working report - Edition 1 - page 4/57

5 10.12 CONFIGURE RINGING TONE AND VOLUME APPENDIX C: ALCATEL-LUCENT ENTERPRISE COMMUNICATION PLATFORM: CONFIGURATION REQUIREMENTS SIP GATEWAY SIP PROXY CODEC OXE DOMAIN SIP USER CONFIGURATION SIP PHONE USER CONFIGURATION APPENDIX D: AAPP MEMBER S ESCALATION PROCESS APPENDIX E: AAPP PROGRAM ALCATEL-LUCENT APPLICATION PARTNER PROGRAM (AAPP) ENTERPRISE.ALCATEL-LUCENT.COM APPENDIX F: AAPP ESCALATION PROCESS INTRODUCTION ESCALATION IN CASE OF A VALID INTER-WORKING REPORT ESCALATION IN ALL OTHER CASES TECHNICAL SUPPORT ACCESS ALE Application Partner Program Inter-working report - Edition 1 - page 5/57

6 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 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 ( in the Application Partner Interworking Reports corner (restricted to Business Partners) - the Application Partner portal ( with free access. ALE Application Partner Program Inter-working report - Edition 1 - page 6/57

7 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 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. ALE Application Partner Program Inter-working report - Edition 1 - page 7/57

8 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. Access to technical support by the Business Partner requires a valid ALE maintenance contract For any request outside the scope of this IWR, ALE offers the On Demand Diagnostic service where assistance will be provided against payment. For details on all cases (3 rd 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 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 will consider that situation as to that where no IWR exists. ALE will handle this situation accordingly (for more details, please refer to Appendix F AAPP Escalation Process ). ALE Application Partner Program Inter-working report - Edition 1 - page 8/57

9 4 Application information Application type Application commercial name: VOIP SIP Phone 9600/9602 MWD, 3300 IP 3302 IP TRM, NDC 2110S E100 IP, E203IP, M203 IP Application version: 9600 IP - CD Single line 9602 IP - CD Two line model 3300 IP - C Single line with Display 3302IPTRM - CT Two line model with disply E100IP - CC Single line E203IP - CD Two Line model NDC2110SN - CD Two Line model M203IP - CD Two line model Interface type: SIP Brief application description: TeleMatrix / Cetis knows that none of today's exciting new telecommunications technologies would be practical unless they were user-friendly, built to last and backed by a strong customer service program. That's why TeleMatrix packages the market's most advanced hospitality and enterprise telephones together with lots of good old-fashioned service and support. The VoIP telephone is used in guest rooms, suites, booths or as administrative phone set in hotel environments. ALE Application Partner Program Inter-working report - Edition 1 - page 9/57

10 Model 9602 MWD ALE Application Partner Program Inter-working report - Edition 1 - page 10/57

11 Model NDC 2110S N Model E100 IP ALE Application Partner Program Inter-working report - Edition 1 - page 11/57

12 Model E203 IP Model E203 IP Model 3302 IP TRM ALE Application Partner Program Inter-working report - Edition 1 - page 12/57

13 ALE Application Partner Program Inter-working report - Edition 1 - page 13/57

14 5 Test environment 5.1 Hardware configuration List main hardware equipments used for testing OmniPCX Entreprise: o Node 1: duplicated CS-V (Call Server Processing Unit- Virtual) o Node 2: Physical nonduplicated Call Server Board o GD (Gateway driver processing Unit) o PRA T2 (ISDN Access) o MIX 2/4/4 (ISDN T0, digital & analog interfaces) o UA digital and analog sets OXE setup OXE Node1 IP addresses / Domain name Attendant No 6666 r12.proservtesting.com OXE Extension Details used for test IP touch and UA extensions 1001 to 1009 SIP users 1202 to 1220 OXE Node2 IP address Domain name Attendant No 6666 Pcs.proservtesting.com OXE Extension Details used for test IP touch and UA extensions 2001 to 2009 ALE Application Partner Program Inter-working report - Edition 1 - page 14/57

15 SIP users 2202 to Software configuration List main softwares used for testing Alcatel-Lucent Enterprise Communication Platform: OmniPCX Enterprise R M d Partner Application : 9600 IP - CD IP - CD (Full Validation was done) 3300 IP - C IPTRM- CT (Full Validation was done) E100IP - CC E203IP - CD NDC2110SN- CD M203IP - CD ALE Application Partner Program Inter-working report - Edition 1 - page 15/57

16 6 Summary of test results 6.1 Summary of main functions supported Features Status Comments Initialization including network configuration Defence SIP registration SIP authentication Voice over IP and RTP codec support Basic calls Local Telephonic Features Date and time display and update OK OK OK OK OK OK OK OK_BUT There are no physical display on following models: 9600, 9602, NDC 2210S, E203IP M203IP, 3300 IP Hotel features Registration as administrative or room set Check-in and check-out OXE Hotel telephonic features Do not disturb Forward Wake-up Voice Mail Call back Conference Enquiry call Broker call Single room / suite management Basic calls Multi-line Multi-occupation OK OK OK OK OK OK OK OK_BUT OK_BUT OK_BUT OK OK OK OK Phone local feature to be used because OXE Suffix are not supported. Phone local feature to be used because OXE Suffix are not supported. Phone local feature to be used because OXE Suffix are not supported. 6.2 Summary of problems None ALE Application Partner Program Inter-working report - Edition 1 - page 16/57

17 6.3 Summary of limitations In Hotel Mode after check-in, unable to leave voice message to a busy user by using the OXE suffix. OXE Conference, Enquiry and Broker call suffixes are not working. Phone local features must be used. OXE voice message suffix is not working. While already in conversation on the two lines, a third incoming call is not correctly processed. In hotel mode when we try to place the call on hold from Cetis sip phone, it is not reflected in the IP touch end. There is not tone or voice is heard until the hold is retrieved. This behavior is not observed in administrative mode, on IP Touch we can observe on display On Hold. Issue reported to Partner but not yet taken in account. 6.4 Notes, remarks Environment setup for Generic Test: o The phone is configured as SIP Extension o It is configured as administrative Environment setup for Hotel Tests o The phone is configured as SIP Extension o It is configured as ROOM + Multi occupancy o And it is CHECKED- IN with a GUEST number in the HOTMENU By default auto provisioning options are enabled in factory reset mode of phone. This causes issues for registering the phone manually. The phone keeps checking for the config file and does not register with details given in the primary register. After disabling the provisioning options the sip register is working fine. In Cordless models, when there is an incoming call the message led is blinking while ringing but it is not blinking in corded models. A beep tone is heard in the CETIS models, when it is rebooting, gets the IP address and till it registering to a server properly. Important configuration adjustment: The dynamic payload number used for the telephonic events (DTMF) has to be adjusted to match between the phone and the OmniPCX Entreprise. Else, calls are impossible. By default, the phone uses 101 and the OmniPCX Entreprise uses 97. So either the OmniPCX Entreprise has to be configured to use 101 or the phone has to be configured to use 97 Full validation was done with models 3302 IP TRM and 9602 IP ALE Application Partner Program Inter-working report - Edition 1 - page 17/57

18 7 Test Result Template The results are presented as indicated in the example below: Test Case Id 1 Test Case N/A OK NOK Comment Test case 1 Action Expected result Test case 2 Action Expected result Test case 3 Action Expected result Test case 4 Action Expected result The application waits for PBX timer or phone set hangs up Relevant only if the CTI interface is a direct CSTA link No indication, no 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 ALE side or on AAPP member side Comment: to be filled in with any relevant comment. Mandatory in case a test has failed especially the reference number of the issue. ALE Application Partner Program Inter-working report - Edition 1 - page 18/57

19 8 Test Results 8.1 Generic tests These tests check the phones behavior for generic SIP phone features like initializations and registrations, audio parameters and voice quality, defenses, basic calls and telephonic features Phone initialization, SIP registration and authentication These tests check that the phones are able to register to the OXE with and without SIP authentication, by using DNS or not, DHCP mode or static IP addressing, 802.1p/Q VLAN tagging and priority. Test Case Id SIP101 SIP102 SIP103 SIP104 Test Case N/A OK NOK Comment SIP set registration to OXE in DHCP mode The phone is configured to get its IP address by DHCP. SIP set registration to OXE in static IP addressing mode The phone is configured to use IP static parameters SIP set registration to OXE using a DNS or alternate proxy The phone is configured to use a domain name as registrar / proxy server address. The DNS IP addresses are the OXE CPU address. In case of alternate proxy possibilities, the main and alternate proxy addresses are the OXE CPU address. Tests are performed when first Call Server is active and then when second Call Server is active SIP set registration to OXE using SIP digest authentication SIP digest authentication is activated on OXE and phone side. Check also that outgoing call is authenticated. ALE Application Partner Program Inter-working report - Edition 1 - page 19/57

20 8.1.2 Audio codecs negotiations These tests check that the phones are using the configured audio parameters (codec). Test Case Id Aud101 Test Case N/A OK NOK Comment The phone is configured to offer G711Alaw, G723 and G729 (in this priority order). The OXE is configured to use G711Alaw. Check that for an incoming and outgoing call, the negotiated codec is G711 A law The phone is configured to offer G711Alaw, G723 and G729 (in this priority order). The OXE is configured to use G723. Aud102 Check that for an incoming and outgoing call, the negotiated codec is G723 The phone is configured to offer G711Alaw, G723 and G729 (in this priority order). The OXE is configured to use G729. Aud103 Check that for an incoming and outgoing call, the negotiated codec is G729 The phone is configured to offer G711Alaw. The OXE is configured to use G711Alaw. Aud104 Check that for an incoming and outgoing call, the negotiated codec is G711 A law The phone is configured to offer G723. The OXE is configured to use G723. Aud105 Check that for an incoming and outgoing call, the negotiated codec is G723. The phone is configured to offer G729. The OXE is configured to use G729. Aud106 Check that for an incoming and outgoing call, the negotiated codec is G729. The phone is configured to offer G722. The OXE is configured to use G722 with 80X8 phones. Aud107 Check that for an incoming and outgoing call, the negotiated codec is G729. Aud108 Repeat previous 6 tests by changing A law to μ law Aud109 Codec selection. The phone and OXE do not have any common codec. For example, the phone is configured to offer G723 and the OXE to use only G729 (use of IP domains). Check that for an incoming and outgoing call is properly rejected. The phones support codec G711A or Mu, G729, G722 and G723. Changed is OXE system parameters. OXE reject call with 488 Not acceptable here. Partner phones send 403 Forbidden ALE Application Partner Program Inter-working report - Edition 1 - page 20/57

21 8.1.3 Defense These tests check the phones defenses against perturbations and OXE Call Servers switch over. Test Case Id DEF10 1 DEF10 2 DEF10 3 DEF10 4 Test Case N/A OK NOK Comment OXE Call Server CPU switch-over while SIP phone in idle. Check the SIP phone behavior after a switch from the OXE main to standby CPU. The phone must be able to make and receive a call after the switch over. OXE Call Server CPU switch-over while SIP phone in conversation with an IPTouch. Check the SIP phone behavior after a switch from the OXE main to standby CPU. The call is still active. After on hook, the phone must be able to make and receive a call after the switch over. OXE Call Server reboot while SIP phone in idle. Check the phone behavior when the OXE Call Server reboots (without standby CPU). As soon as the Call Server is running again, the phone is able to make and receive a call. OXE Call Server reboot while SIP phone in conversation with an IPTouch. Check the phone behavior when the OXE Call Server reboots (without standby CPU). The call is released. As soon as the Call Server is running again, the phone is able to make and receive a call. ALE Application Partner Program Inter-working report - Edition 1 - page 21/57

22 8.1.4 Basic call These tests check the phone behavior during basic incoming and outgoing calls from and to different kind of phone set types (SIP, IPTouch, UA) with different call releases (during ringing, by caller, by callee) and with or without a second incoming call. Test Case Id Test Case N/A OK NOK Comment Call from and to a SIP phone. CAL10 1 CAL10 2 CAL10 3 CAL10 4 CAL10 5 CAL10 6 CAL10 7 CAL10 8 CAL10 9 CAL11 0 CAL11 1 CAL11 2 The phone calls a SIP phone. The phone is called by a SIP phone. In both cases, check the display and audio during all steps (dialing, ring back tone, conversation, and release). Call from and to an IPTouch. Same as 1 but with an IPTouch. Call from and to an UA phone. Same as 1 but with an UA phone. Incoming call released by the caller during ringing. The caller releases the incoming call to the phone before the callee takes the call. Outgoing call released by the caller during ringing. The caller releases the outgoing call from the phone before the callee takes the call. Incoming call rejected by the callee during ringing. The callee rejects the incoming call to the phone during ringing. Outgoing call rejected by the callee during ringing. The callee rejects the outgoing call from the phone during ringing. Call released by the phone. The phone releases the call after a conversation period. Call released by the other phone. The other phone releases the call after a conversation period. Call from and to an external number (T0/T2) Call is properly established. Call from and to an attendant Call is properly established. Incoming external call (T0/T2 for example) to an attendant phone set which transfers the call to the phone. Transfer is done while the phone is ringing but also after this one has picked up the call (using the attendant soft key or going on hook). No possibility to reject a call on this phone. ALE Application Partner Program Inter-working report - Edition 1 - page 22/57

23 Test Case Id CAL11 3 CAL11 4 Test Case N/A OK NOK Comment Call is properly established. Outgoing call from a phone to an attendant with transfers to an external call (T0/T2 for example). Call is properly established. Dialing break The phone starts dialing another phone number. Before the end, the dialing is stopped. Check that the phone comes back to idle state after the timeout expires Local telephonic features These tests check the phone behavior during phone local telephonic feature use like forward, on hold, transfer, voice mail interactions, conference. These features are either activated through the phone web manager or with phone keys (if available). Test Case Id TEL101 Test Case N/A OK NOK Comment Immediate forward to another phone. The phone is forwarded to another phone. Call The phone and check that the call is presented on the third phone and can be taken by this one. Forward on no answer to another phone. TEL102 The phone is forwarded on no answer to another phone. Call the phone and check that the call is presented on analog phone. Do not take the call and wait for the call to be presented on the third phone. Take the call on the third phone. Check also the call can be picked up before the call is forwarded. Forward on busy to another phone. TEL103 The phone is forwarded on busy to another phone. While the phone is already in conversation, call The phone and check that the call is presented. Do not take the call and wait for the call to be presented on the third phone. Take the call on the third phone. TEL104 Forward on busy / no answer to another phone. The phone is forwarded on busy / no answer to another phone. While the phone is already in conversation, call The phone and check that the call is presented. Do not take the call and wait for the call to be presented on the third phone. Take the call on the third phone. ALE Application Partner Program Inter-working report - Edition 1 - page 23/57 Not possible to enable forward on busy and forward no answer same time.

24 Test Case Id Test Case N/A OK NOK Comment Call The phone and check that the call is presented. Do not take the call and wait for the call to be presented on the third phone. Take the call on the third phone. Check also the call can be picked up before the call is forwarded. Phone puts call on-hold. TEL105 TEL106 TEL107 TEL108 TEL109 TEL110 TEL111 The phone is in conversation with another phone. This conversation is put on-hold. Check the display and on-hold music on this phone. Check also the display and audio signalization on The phone. Check the conversation can be retrieved. Phone is put on-hold. The phone is in conversation with another phone. The other phone puts this conversation on-hold. Check the display and on-hold music on The phone. Check the conversation can be retrieved. Broker call. The phone has two active conversations and switches from one to another. Check the display and on-hold music on the phones. Check also the display and audio signalization on the phone. Transfer in conversation. The phone has two active conversations and transfers the first to the second. Check the new conversation between the two other phones is successful and also the phone display (signalization of the transfer and back to idle state). Transfer during ringing. The phone has one active conversation and another one in ringing step. Before the second callee takes the call, The phone transfers its first call to this second callee. Check the new conversation between the two other phones is successful and also The phone display (signalization of the transfer and back to idle state). Conference. The phone has two active conversations and initiates a conference. Check the new conversation between the three parties is successful (audio and signalization). Do Not Disturb. On The phone the local feature (if exists) Do not Disturb is activated. When calling this set, the call is not presented on the phone. Transfer option available only with specific model 3302IP TRM Conference can be established only with multiline phones 9602IP NDC2110 S E203IP 3302IP TRM On The phone the Do not Disturb is deactivated. When calling this set, the call is presented on the ALE Application Partner Program Inter-working report - Edition 1 - page 24/57

25 Test Case Id Test Case N/A OK NOK Comment phone and can be picked up. TEL112 Wake Up. On The phone the local feature (if exists) Wake Up is activated. When the wake up time arrives, the phone rings. When the picked up, the voice guide is played. Test also with The phone already in conversation when the wakeup time arrives. On The phone the Wake Up is activated then deactivated. When the previous wake up time arrives, nothing appends on the phone set. No local feature available in Telematrix Phone Other features These tests check the phone behavior while using features like STP (date and time display). Test Case Id OTH10 1 Test Case N/A OK NOK Comment Date and time display using Network Time Protocol. The SIP phone is configured to retrieve the date and time from a NTP server. Check the phone retrieves the information and displays it IP TRM 3300 IP E100 IP Only in the above models date can be seen as these models have physical display. ALE Application Partner Program Inter-working report - Edition 1 - page 25/57

26 8.2 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 Provisioning These tests check the phone provisioning as a room, suite, administrative or booth set. Test Case Id Test Case N/A OK NOK Comment Declare the SIP phone as a hotel room set HOT10 1 HOT10 2 HOT10 3 In the OXE configuration, the SIP set is declared as a hotel room set Or 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. Declare the SIP phone as a hotel administrative set In the OXE configuration, the SIP set is declared as a hotel administrative set Or The SIP set is configured (user and password) with the parameters of an already declared SIP hotel administrative set The phone registers correctly to the OXE. Declare the SIP phone as a hotel booth set In the OXE configuration, the SIP set is declared as a hotel booth set Or The SIP set is configured (user and password) with the parameters of an already declared SIP hotel booth set The phone registers correctly to the OXE. Configured as Room in Room management mode. Configured as Administrative Configured as House in user management ALE Application Partner Program Inter-working report - Edition 1 - page 26/57

27 8.2.2 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 Case Id CHK10 1 CHK10 2 CHK10 3 Test Case N/A OK NOK Comment 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 the SIP phone room Or The SIP set is configured (user and password) with the parameters of an already declared SIP hotel room set in which a client has already checked-in. A client checks in as VIP in a room containing the SIP phone Same as above but with the VIP 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 containing the SIP phone In the OXE hotel menu (hotmenu), a check-out is done for the client using the SIP phone room Do not disturb These tests check the phone behavior in case of "Do not disturb" activation / deactivation (on the phone itself or from an administrative phone). Test Case Id DND10 1 DND10 2 DND10 3 Test Case N/A OK NOK Comment Do Not Disturb is activated on the room SIP phone On the room SIP phone the Do not Disturb is activated thanks to the prefix 42 (and then personal password) When calling this set, the call is not presented on the phone. Do Not Disturb is deactivated on the room SIP phone On the room SIP phone the Do not Disturb is deactivated thanks to the prefix 42 When calling this set, the call is presented on the phone and can be picked up. Do Not Disturb is activated on the suite master SIP phone On the suite master SIP phone the Do not Disturb is activated thanks to the prefix 586 When calling this guest (call to the guest number ALE Application Partner Program Inter-working report - Edition 1 - page 27/57

28 and call to the main and slave suite phones), the call is not presented on the phones. Same behavior when calling a slave SIP phone set. Do Not Disturb is deactivated on the suite master SIP phone DND10 4 On the suite master SIP phone the Do not Disturb is deactivated thanks to the prefix 586 When calling this guest, the call is presented on the main suite phone and can be picked up. Same behavior when calling a slave SIP phone set Forward These tests check the phone behavior in case of "Forward" activation / deactivation (on the phone itself or from an administrative phone), to an administrative set or to the voice mail. To allow the available forward functions it is required to have a checked-in guest with a voic . Test Case Id FWD10 1 FWD10 2 FWD10 3 FWD10 4 FWD10 5 Test Case N/A OK NOK Comment Immediate forward is activated on the room SIP phone On the room SIP phone the Forward is activated thanks to the prefix 51 (immediate forward). Forward destination is an administrative SIP set or the voice mail. When calling this set, the call is not presented on the phone but forwarded to the administrative phone or voice mail. Forward is deactivated on the room SIP phone On the room SIP phone the Forward is deactivated thanks to the prefix 41. When calling this set, the call is presented on the phone and can be picked up. Call to a phone, which is forwarded to another phone. The phone calls a phone forwarded to a third phone. The third phone takes the call and the conversation is established. Forward on no answer to another phone. OXE Prefix: 53 The phone is forwarded on no answer to another phone. Call the phone and check that the call is presented on third phone. Do not take the call and wait for the call to be presented on the third phone. Take the call on the third phone. Check also the call can be picked up before the call is forwarded. Forward on busy to another phone. Prefix: 52 The phone is forwarded on busy to another phone. While the phone is already in conversation, call The phone and check that the call is presented. Do not take the call and wait for the call to be presented on ALE Application Partner Program Inter-working report - Edition 1 - page 28/57

29 Test Case Id Test Case N/A OK NOK Comment the third phone. Take the call on the third phone. Forward on busy / no answer to another phone. OXE Prefix: 54 The phone is forwarded on busy / no answer to another phone. FWD10 6 While the phone is already in conversation, call The phone and check that the call is presented. Do not take the call and wait for the call to be presented on the third phone. Take the call on the third phone. Call The phone and check that the call is presented. Do not take the call and wait for the call to be presented on the third phone. Take the call on the third phone. Check also the call can be picked up before the call is forwarded Voice mail These tests check the phone behavior when interworking with the OXE 4645 voic . In order to do these tests, the Room has to be checked in. If the room isn t checked-in there is no voic available. In the User parameters, make sure that before the Check-in there is no Voic number. Test Case Id VMA10 1 VMA10 2 VMA10 3 VMA10 4 Test Case N/A OK NOK Comment Voice mail message signalization. The voice mail (OXE 4645) number is configured in the phone. Call The 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 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. Voice mail message deposit. The phone calls another phone forwarded to the voice mail. He leaves a message. Check the interaction between the phone and voice mail. Listen to the message from the other phone. Voice mail message deposit. Suffix: 8 Check the can leave a voice message to a phone which is not answering or already in conversation thanks to OXE suffix. Message Light LED is blinking while a message was dropped in box. ALE Application Partner Program Inter-working report - Edition 1 - page 29/57

30 8.2.6 Wake-up Calls 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 Case Id WUC101 WUC102 WUC103 Test Case N/A OK NOK Comment Wake Up is activated on the room SIP phone On the room SIP phone the Wake Up 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 On the room SIP phone the Wake Up 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 When the wake up time arrives, the 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 WUC104 On the suite master SIP phone the suite wake Up is activated thanks to the prefix 506 When the wake up time arrives, only 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. Test also the activation from slave suite phones. Behavior is the same. Suite wake Up is deactivated on a suite SIP phone WUC105 On the suite master SIP phone the Wake Up is deactivated thanks to the prefix 585 When the previous wake up time arrives, nothing appends on the phone sets (master and slaves). Test also the deactivation from slave suite phones. Behavior is the same. ALE Application Partner Program Inter-working report - Edition 1 - page 30/57

31 8.2.7 Change room state These tests check the phone can change the room state (as a room or administrative set). Test Case Id STA10 1 Test Case N/A OK NOK Comment 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 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) Mini bar These tests check the phone can change the "mini bar" state (as a room or administrative set). Test Case Id Test Case N/A OK NOK Comment The min-bar state is changed on the room SIP phone set. BAR10 1 On the room SIP phone the mini-bar state is changed thanks to the prefix 588 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. Prefix is sent without codes behind Hotel Calls These tests check the phone behavior during calls between rooms, from and to an administrative set, external outgoing call. It also checks the transfer feature. Test Case Id HCA10 1 HCA10 2 Test Case N/A OK NOK Comment SIP room phone set to another SIP room phone set (different rooms). Using guest and room numbers. SIP room phone to an administrative phone set and vice-versa. ALE Application Partner Program Inter-working report - Edition 1 - page 31/57

32 Test Case Id Test Case N/A OK NOK Comment Using guest and room numbers. HCA10 3 HCA10 4 HCA10 5 HCA10 6 HCA10 7 HCA10 8 SIP room phone set to an external number (T0/T2 for example) and vice versa SIP suite phone set to an external number (T0/T2 for example) and vice versa. Try with the master and several slave phone sets. SIP booth phone set to an external number (T0/T2 for example) and vice versa. SIP room phone to an attendant and vice versa. Call is properly established. Incoming external call (T0/T2 for example) to an attendant phone set which transfers the call to the SIP room phone. Transfer is done while the SIP phone is ringing but also after this one has picked up the call (using the attendant soft key or going on hook). Call is properly established. Outgoing call from a SIP phone to an attendant with transfers to an external call (T0/T2 for example). Call is properly established. Call back on no answer / busy. Suffix: 5 HCA10 9 HCA11 0 HCA11 1 HCA11 2 The phone calls another phone already in conversation (or does not answer check both). The phone set uses the call back suffix to be recalled. Check the call back query is taken into account and processed. Conference. Suffix: 3 Check the phone can establish a three party conference thanks to OXE suffix. Broker call. Suffix: 1 Check the phone can switch between two calls thanks to OXE suffix. Enquiry call. Suffix: 2 Check the phone can make a second call while already in conversation thanks to OXE suffix. OXE Suffix not supported. Phone local feature to be used. OXE Suffix not supported. Phone local feature to be used. OXE Suffix not supported. Phone local feature to be used. ALE Application Partner Program Inter-working report - Edition 1 - page 32/57

33 Multi-line These tests check the phone behavior during multi-line calls from and to different kind of phone set types (SIP, IPTouch, UA). Room and suite phones are tested. We used the multi-line feature of the partner phones and not the one from Pbx. Test Case Id MLE10 1 MLE10 2 MLE10 3 Test Case N/A OK NOK Comment Incoming call to a room phone while already in conversation The phone (only phone of the room) is already in conversation and receives a new incoming call. Check the display (new call presentation) and audio (new call signalization). Check this call can be taken (and that the first one is put on hold). Check the phone can switch between the two calls (and that the other parties are put on hold). Outgoing call from a room phone while already in conversation The phone (only phone in the room) is already in conversation and makes a new outgoing call. Check the display (new call presentation) and audio (new call signalization). Check this call can be taken (and that the first one is put on hold). Check the phone can switch between the two calls (and that the other parties are put on hold). Incoming call to a room phone while already in conversation on the two lines The phone (only phone in the room) is already in conversation on its two lines and receives a new incoming call. Check the display (new call presentation) and audio (new call signalization). Incoming call to suite phone while already in conversation MLE10 4 MLE10 5 The phone (main phone of a suite) is already in conversation and receives a new incoming call. Check the display (new call presentation) and audio (new call signalization). Check this call can be taken (and that the first one is put on hold). Check that the slave phones stop ringing. Check the phone can switch between the two calls (and that the other parties are put on hold). Outgoing call from a suite phone while already in conversation The phone (main phone of a suite) is already in conversation and makes a new outgoing call. Check the display (new call presentation) and audio (new call signalization). ALE Application Partner Program Inter-working report - Edition 1 - page 33/57

34 Test Case Id MLE10 6 MLE10 7 MLE10 8 MLE10 9 Test Case N/A OK NOK Comment Check this call can be taken (and that the first one is put on hold). Check the phone can switch between the two calls (and that the other parties are put on hold). Incoming call to a suite phone while already in conversation on the two lines The phone (main phone of a suite) is already in conversation on its two lines and receives a new incoming call. Check the display (new call presentation) and audio (new call signalization). Check that the slave phones ring and can take the call. Repeat steps 4 to 6 but this time the SIP phone is a suite slave phone. Outgoing call from a room phone to another room phone which is already in conversation The phone calls another room phone (only phone in the room) which is already in conversation and, thanks to Busy Camp On feature (suffix 6), enforce the called to phone to ring. Check the display (new call presentation) and audio (new call signalization). Check this call can be taken (and that the first one is put on hold). Check the phone can switch between the two calls (and that the other parties are put on hold). Repeat previous steps but, this time, use guest number (and not room number) when calling to a room Multi-occupation These tests check the phone behavior when several guests are hosted in the same room. Test Case Id MOC101 Test Case N/A OK NOK Comment SIP set registration to OXE The phone successfully registers to the OXE Incoming call to the room phone number MOC102 MOC103 MOC104 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 Another phone (IPTouch) calls the first guest phone number. The call can be picked up and is successfully established. Incoming call to the second guest phone number Another phone (IPTouch) calls the second guest phone number. The call can be picked up and is ALE Application Partner Program Inter-working report - Edition 1 - page 34/57

35 successfully established. Outgoing call by the first guest MOC105 MOC106 In case of an external call (to a PSTN 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 In case of an external call (to a PSTN user for example), the second guest makes an outgoing call using his guest ID. The call is successfully established Additional Tests. These tests check the good behavior in hotel suite mode. Test Case Id ADD10 1 ADD10 2 Test Case N/A OK NOK Comment For a suite with several phones, check behavior when there is an incoming call (to the suite phone 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 like forward, do not disturb, wake up time) ALE Application Partner Program Inter-working report - Edition 1 - page 35/57

36 9 Appendix A : AAPP member s Application description Cetis manufactures multiple suites of cordless, corded, and trimline, compact-form factor IP phones for the hospitality sector. These SIP phones interact with Alcatel as a basic SIP phone. While each model has slight variations to accommodate physical and logical differences (such as number of lines/display etc), the core firmware is the same for all models. All models also use the same DSPG chipset. Cordless models have additional hardware for DECT 6.0. The main protocols in action and web GUI configuration sections: SIP/RTP/DTMF to make VoIP calls and interact during the call. Can be configured under Primary Register TFTP/HTTP/HTTPS to facilitate provisioning. HTTP is used to access the web configuration. Can be configured under Provisioning Standard network protocols NTP, DNS, DHCP, IP/TCP, etc. Does not support IPv6. Other: The common mechanism for Cetis phones to receive configuration is via a config ID prompt at first boot (<ConfigID>.cetis.cfg file). This allows the user to create configuration files based off of room number/extension and then enter that room number to tell the phone to provision itself from provisioning server via option 66 or other provisioning methods. Cetis VoIP phones also support provisioning via MAC address configuration files (<macaddress>.cetis.cfg). All Cetis phones are IEEE 802.3af compliant. E-series cordless and corded model (excluding trimline models) come equipped with 1 USB for charging devices. MWI indicator key is also used as the speed-dial for message retrieval ALE Application Partner Program Inter-working report - Edition 1 - page 36/57

37 10 Appendix B: Configuration requirements of the AAPP member s application 10.1 VOIP Series Configuration This section describes the configuration settings required for the Cetis VOIP Series integration with Alcatel OxE, primarily focusing on the SIP interface configuration. The Cetis VOIP Series configuration settings identified in this section have been derived and verified through interoperability testing with Alcatel. For configuration details not covered in this section, see the VoIP User Guides: found at for Cetis VOIP Series. Configuration Method Cetis VOIP Series phones can be configured via configuration files, webui, and limited keypad codes. It is recommended that configuration files are stored on a provisioning server (TFTP/HTTP) for mass installations and quick adds/replaces by hotel staff. The network must have DHCP option 66 or option 60/43 enabled on the network for quick configuration. Otherwise, configuration will require some manual configuration via the keypad to obtain a static IP address and then via the webui to obtain the correct provisioning/network information. See the VoIP User Guide for Cetis VOIP Series for manual configuration of network and provisioning settings. Visit support.cetis.com/portal/voip for additional support and documentation. Cetis VOIP Series phones provide for several types of configuration files as described in the table below. Configuration Files Cetis VOIP Series Configuration Level Description Files <Model Prefix>-3.x.x-xxx.bin (Current GA ) Example: Cetis corded two-line E- series model would be CC bin System This file contains the device firmware load. All models share the same base SIP stack. Firmware has variations are to accommodate display, line, and other physical differences not shared by models. This is represented by a model prefix such as CC1, CD1, etc. firmware.cetis.txt System This file contains firmware upgrade/downgrade information for phones configured to look for this file(on by default). <ConfigID>.cetis.cfg Example: 1234.cetis.cfg Subscriber This file contains configurable parameters that apply to an individual device in a deployment. The ConfigID is a way of naming a file by extension or room number allowing the user to create configuration files, place them on a configuration server, and then enter in the unique ConfigID to configure a phone. This requires no barcode scanning (as in the case of MAC files) and allows housekeeping to easily replace broken phones. These settings can be configured in the webui/config file under ALE Application Partner Program Inter-working report - Edition 1 - page 37/57

38 Cetis VOIP Series Configuration Files <macadress>.cetis.cfg Example: 0019f30f51b2.cetis.cfg Level Subscriber Description Provisioning. This file is used alternatively to a ConfigID file (described above). Instead of the user needing to enter the ConfigID during installation, the user can create configuration files with the phone s MAC address. If this is the case then the user needs to disable ConfigID file requests so both MAC address files and ConfigID files are not requested. To allow the customer to choose a method, Cetis phones have both methods enabled by default. Mac address file requests are attempted first. These settings can be configured in the webui/config file under Provisioning. Configure Network Settings This section describes the network related settings on Cetis VOIP Series Configure IP address The IP address can either be configured via DHCP or statically. The default is DHCP. Static configuration can be achieved from the keypad of the phone for networks where DHCP is not available with a series of key codes: 1) Press # after initial boot to bypass ConfigID prompt. ConfigID prompt is a method of retrieving a configuration file, but first network access must be obtained. The prompt will be indicated with a doodle-doodle jingle and pulsating LED lights. 2) Press **73*<keypad password (default 123)># to change the network mode from DHCP to Static. 3) Press **74*<keypad password (default 123)>*<IP address># to set the static IP address. 4) Press **76*<keypad password (default 123)>*<subnet mask># to set the subnet mask. 5) Press **49*<<keypad password (default 123)>*<gateway IP address># to set the gateway IP address. The gateway address must be in the same network as static IP address. 6) Power cycle phone to apply new network settings Access the webui The Cetis VOIP Series IP address can be retrieved from the phone by following below procedure: 1) On the phone, press # to bypass the ConfigID prompt. 2) Press **47# to hear and/or display the IP address of the phone. ALE Application Partner Program Inter-working report - Edition 1 - page 38/57

39 After learning the phone s IP address, access the phone webui through address of the phone> from a web browser. Enter the credentials to login. The default login username is admin, the default password is admin. Press Login on the screen or Enter key of the computer will display the Home screen. This screen has the current status of the phone. ALE Application Partner Program Inter-working report - Edition 1 - page 39/57

40 10.4 Configure DNS Phones using DHCP should receive DNS server information via DHCP. For a statically addressed phone, configure DNS servers through WAN Settings -> Basic Settings screen after login into the phone. Press Apply after the setting is completed. This will prompt for a reboot. The prompt may be ignored until other configuration is filled in. But make sure to reboot the phone to apply the DNS settings after other configuration is complete. The type of DNS record can also be configured. Do this by going to Primary Register -> Protocol Control. Choose the appropriate DNS settings. ALE Application Partner Program Inter-working report - Edition 1 - page 40/57

41 After the configuration is completed, press Apply to reboot the phone Configure NTP NTP and other time settings can be changed under Time Setting. To use the phone s SNTP service (simplified NTP), choose the Enable radial button, and fill in the correct server address. By default, a public NTP server pool is filled in: Next, the correct Time Zone for the region where the phone is installed should be used. This adjusts the UTC time from the NTP server to the local time of the client. If the phone is in a region with Daylight Savings Time (DST), then configure DST under Daylight Savings Settings according to region as well: Press Apply after the setting is completed. If DST is set to Off on the above screen, the phone display time can be toggled forward/backward an hour by pressing **37# from the phone itself Configure Transport Protocols Configure the transport protocol used for SIP under Primary Register -> Protocol Control section. The default transport protocol is UDP: ALE Application Partner Program Inter-working report - Edition 1 - page 41/57

42 Choose either TCP or UDP. TLS has not been verified to during interoperability test. Press Apply after the setting is completed Configure SIP Interface Settings This section describes the SIP configuration items that are required for each Cetis VOIP Series phone to work with and SIP server Configure SIP Registrar Settings Configure the registrar under Primary Register -> Register Server section. Enable the registration service via the Use Service drop-down field. Fill in Display Name field (optional). Fill in SIP user in User Name field. Fill in authentication user name in Authorization User Name field Fill in authentication password in Password. The authentication password must match the SIP authentication password configured on the server. Enter IP or FQDN of the Alcatel server in Register Server Address field. If using an FQDN, verify DNS is configured correctly. If using an outbound proxy, enter the FQDN or IP address of proxy in Outbound Proxy field. Change SIP registration refreshes by changing the Register Expire field. This timer will be overridden by the timer negotiated between the phone and the Alcatel registration server and can usually be left at default. Press Apply after the setting is completed. Configure MWI and Other SIP Settings This section describes how to configure MWI, Session Timer, Session Audit, DTMF, Audio Settings, and Prack (100rel) for the Cetis VOIP Series phone. Configure MWI Enable or disable Message Waiting Indication (MWI) from Primary Register -> Protocol Control section. The MWI is enabled by default. ALE Application Partner Program Inter-working report - Edition 1 - page 42/57

43 If Subscribe MWI is enabled, the Subscribe Expire timer can be changed from below screen. This timer will be overridden by the timer negotiated between the phone and Alcatel and is usually left at default. Press Apply after the setting is completed. Configure DTMF Change DTMF mode from the same screen as MWI. The default DTMF mode is RFC2833. Cetis VOIP Series will automatically fallback to Inband method when RFC2833 is not supported. Press Apply after the setting is completed. Enable Support Update Method to allow UPDATE message is used during Session Timer and Session Timer. This step is necessary for correct function of both Session Timer and Session Audit. No additional step is needed if using Session Audit. Press Apply after the setting is completed. Configure PRACK PRACK can be enabled or disabled from the same screen as MWI. Press Apply after the setting is completed Configure Codecs Codec settings can be changed from Audio Settings -> Codec Settings screen. Press Apply after the setting is completed Configure Service Settings This section describes how to configure Memory/Programmable keys, Call forwarding, and Dial Plan on Cetis VOIP Series. Configure Memory/Programmable keys Configure Memory/Programmable keys under Call Features -> Programmable Keys & MWI Number. To configure a memory key, simply select Memory in the drop-down box and enter the correct number: Program other functions by selecting the appropriate setting from drop-down list. ALE Application Partner Program Inter-working report - Edition 1 - page 43/57

44 Fill in MWI Number with voice portal number to allow message retrieving from MWI button. The example s4221pp4221# allows voic number 4221 is automatically dialed and voic password 4221 is automatically inputted after a long pause (pp) when MWI button is pressed. Press Apply after the setting is completed Call Forwarding Configure call forwarding under Call Features section Fill in the forward number. If forward type is Busy Forward, disable Call Waiting feature. Press Apply after the setting is completed. ALE Application Partner Program Inter-working report - Edition 1 - page 44/57

45 Configure Dialing Plan Configure the Dial Plan under the Dialing Rules section. An example for North America dial plan is shown below. Please program to preference Configure Ringing Tone and Volume Configure Ring tones and volume settings under Audio Settings -> Sound and Volume Control section. Press Apply after the setting is completed.. ALE Application Partner Program Inter-working report - Edition 1 - page 45/57

46 11 Appendix C: Alcatel-Lucent Enterprise Communication Platform: configuration requirements Launch OXE configuration application (console with command MGR or GUI with OmniVista8770) 11.1 SIP gateway 11.2 SIP Proxy ALE Application Partner Program Inter-working report - Edition 1 - page 46/57

47 11.3 Codec A Law/ Mu Law We can set the codec to G729 in the below path. Select: System > Other System Param. > Compression Parameters Compression Type Select: G OXE domain ALE Application Partner Program Inter-working report - Edition 1 - page 47/57

48 11.5 SIP user configuration Hotel Set configuration Guest Management ALE Application Partner Program Inter-working report - Edition 1 - page 48/57

49 Room Management Administrative set configuration For tracing purpose we can use below commands in OXE (101) etesting_b> motortrace c motortrace (v5.2.0) verbosity = 0037b524 sipmotor trace-level set c (data dump). ALE Application Partner Program Inter-working report - Edition 1 - page 49/57

50 11.6 SIP Phone user configuration. Please check for the SIP licenses before the SIP phones are installed. Uses Spadmin and check lock # 177. ALE Application Partner Program Inter-working report - Edition 1 - page 50/57

51 12 Appendix D: AAPP member s escalation process Contact name: Vincent Navarrete Phone: Fax: ALE Application Partner Program Inter-working report - Edition 1 - page 51/57

52 13 Appendix E: AAPP program 13.1 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 thirdparty applications that complement Alcatel-Lucent Enterprise's product family. ALE 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 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. ALE Application Partner Program Inter-working report - Edition 1 - page 52/57

53 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 Enterprise.Alcatel-Lucent.com You can access the Alcatel-Lucent Enterprise website at this URL: ALE Application Partner Program Inter-working report - Edition 1 - page 53/57

54 14 Appendix F: AAPP Escalation process 14.1 Introduction The purpose of this appendix is to define the escalation process to be applied by the ALE Business Partners when facing a problem with the solution certified in this document. The principle is that ALE 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 and the Application Partner, are engaged as following: (*) The Application Partner Business Partner can be a Third-Party company or the ALE Business Partner itself 14.2 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 and the Application Partner, are engaged: Case 1: the responsibility can be established 100% on ALE side. In that case, the problem must be escalated by the ALE Business Partner to the ALE Support Center using the standard process: open a ticket (eservice Request esr) ALE Application Partner Program Inter-working report - Edition 1 - page 54/57