ID Features Tested Case Title Description Call Component Flow Status Defects UC713L.ANA.001 Unified Communications Manager

Transcription

1 Analog System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713L.ANA.001 Manager IP to analog calls Verify that calls from SFO-ORD IP Phone to SFO-ORD Analog phone are successful. UC713L.ANA.001 Analog Calls between IP Phone and Analog Phone Verify that calls from SFO-ORD IP Phone to SFO-ORD Analog phone are successful 1 of 214

2 Cisco Emergency Responder System Test Results for IP Telephony: Cisco System Release 7.1(3) SR60.CER Emergency Failed CSCtb04572 Responder System Reliability When Single Emergency Responder Server within the Server Group Fails - Fallback Verify to ensure that there is no single point of failure in the Emergency Responder Server Group. SR60.CER SR60.CER OXN51.CER.004 Emergency Responder Emergency Responder Emergency Responder IP Phones:PSAP Callback the E911 Caller Over SIP VoIP Protocol IP Phones and IP Phone 6921/6941/6961: Track Current Location of IP Phone and If E911 Call is Routed to Nearest PSAP Emergency Responder PSAP Able to Reach IP Communicator after 911 Call Verify that PSAP can call back the E911 caller. Verify if Emergency Responder can track current location of IP Phone and if E911 calls by users can get routed to nearest PSAP. Verifies that Cisco IP Communicator can be called from Cisco Emergency Responder after Cisco IP Communicator placed a 911 call. PSAP->PSTN->Gateway- Manager->Emergency Responder-> > IP Phone IP Phone-> >Emergency Responder- Manager->Gateway->PSTN- >PSAP IP Communicator-> >Emergency Responder GB40.ER.005 GB40.ER.003 Emergency Responder Emergency Responder IP-to-IP Intra-cluster Emergency Responder Calls After Phone Relocation IP-IP Inter-cluster Emergency Responder for Call Transfer Lines Verify that Emergency Responder calls can be made after the phone has been relocated to another access switch. Verify that call transfer can utilize Emergency Responder for E911 calls. 2 of 214

3 Cisco Emergency Responder System Test Results for IP Telephony: Cisco System Release 7.1(3) GB40.ER.002 Emergency Responder IP-to-IP Intra-cluster Emergency Responder for Shared Lines Verify that shared lines phones can utilize Emergency Responder for E911 calls. GB31.ER.285 GB31.ER.282 Emergency Responder Emergency Responder IP-IP Intra-cluster Multiple Emergency Responder Calls IP-IP Intra-cluster Emergency Responder Calls Using Extension Mobility Verify IP-IP intra-cluster multiple Emergency Responder calls. Verify by making IP-IP intracluster Emergency Responder calls using Extension Mobility. GB31.ER.276 Emergency Responder Emergency Responder Emergency Responder phone Phone Discovery and IP-discovery and IP-IP Emergency IP Emergency Responder. Responder 3 of 214

4 Gateways System Test Results for IP Telephony: Cisco System Release 7.1(3) UCS712IF.GTW.001 Supplementary Verify RFC2833 On Call Verify RFC 2833 works on a Features on VG204 and FXS From Manager controlled VG202 port. cards Manager VG202 Phone The call is established from To A PSTN Phone. Manager controlled VG202 phone to a PSTN phone. UC713L.GTW.105 Gateways IP to PSTN to IP Calls Verify that calls from AZO IP Phone to AZO IP Phone over PSTN are successful UC713IL.GTW.105 Gateways Calls Between IP Phones Over PSTN UC713IL.GTW.003 Gateways Calls Between IP Phone and PSTN Phone Verify that calls between IP Phones over PSTN are successful. Verify that calls between IP Phone and PSTN Phone are successful. 4 of 214

5 Inter-cluster Trunk System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713IL.ICT.002 Inter-cluster Trunk Calls Between IP Phones Over Inter- Cluster Trunk Verify that calls between IP Phones over Inter-Cluster Trunks are successful. 5 of 214

6 Interoperability System Test Results for IP Telephony: Cisco System Release 7.1(3) SR60.E.102 Interoperability Interoperability Incoming Call through H.323 Trunk Establish an incoming call from 4.2 to 6.0 through H.323 Trunk Transfer the call to another IP Phone 6 of 214

7 IP Communicator System Test Results for IP Telephony: Cisco System Release 7.1(3) UC712IF.IPC.004 IP Communicator Authenticate SCCP IP Communicator, IP Phone and Joining Secure Meet-Me Conference Using G.722 Verify by placing an authenticated SIP IP Communicator call and join a secure Meet-Me conference using the G.722 codec. IP Communicator-> >Conference UC712IF.IPC.005 IP Communicator Authenticated SIP IP Communicator Call Over Secured non- Gatekeeper Controlled H.323 ICT, Including a Call Park Verify by placing an authenticated SIP IP Communicator call to a IP Phone 6921/6941/6961 over a secured non-gk controlled H.323 ICT. The IP Phone 6921/6941/6961 the parks the call which is retrieved by a third phone in the same cluster. IP Communicator-> >ICT- Manager-> IP Phone B- Manager-> IP Phone C UC712IF.IPC.007 IP Communicator Use IP Communicator to Make Authenticated Call through TLS Proxy Verify by using IP Communicator to make authenticated call through TLS Proxy. 7 of 214

8 Network Management System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713IL.NME.003 Network Management Service Statistics Manager support to devices Verify the ability to provide visibility into and usability of key metrics such as call volume, service availability, call quality, resource utilization, and capacity for a large enterprise test bed of up to 30,000 devices. UC713IL.NME.002 Network Management Service Monitor Support to Devices Verify comprehensive voice quality measurements through the combination of Cisco 1040 Sensors and Cisco VTQ (Voice Transmission Quality) and alert generation sent to an upstream application ( Operations Manager) when an MOS threshold is violated. Verification is accomplished using test beds supporting up to 30,000 devices. UC713IL.NME.001 OXN51.NME.001 Network Management Alerts and Activity Display Operations Manager Support to Devices and Users Alerts and Activity Display in Service Level View Verifies functionality of Operations Manager during use with a large scale test bed supporting up to 30,000 devices and users. Verify the status of the Manager alert is represented by: Active - New alert generated, ACK - Acknowledged by user, Clear - Cleared. 8 of 214

9 Network Management System Test Results for IP Telephony: Cisco System Release 7.1(3) GB40.NM.007 Network Management GB40.NM.005 GB40.NM.004 Network Management Network Management ITM Will Display Phones in SRST Mode and Generate Related Alerts Use ITM to Do Discovery of All New Phones and Create Scheduled Discovery Verify by importing Site 6,7 and 8 SRST information and identify the SRST components for ITM to display SRST phones and identify alerts. Verify that you can discovery all the phones in GB4.0 test bed across the WAN and inter site. Phone Configuration Verify by configuring a study of Using ITM Configuration phones using ITM configuration Tool tool. 9 of 214

10 PhoneSuite System Test Results for IP Telephony: Cisco System Release 7.1(3) UC712IF.UCW.100 PhoneSuite Click to Dial from Microsoft Internet Explorer Verify if phone numbers within MS Internet Explorer can be dialed using single Click. 10 of 214

11 QSIG System Test Results for IP Telephony: Cisco System Release 7.1(3) UC712EF.QSG.003 QSIG per trunk Inter PBX Call Via Manager Clusters Verify inter PBX call via Manager clusters. PBX Phone 1->QSIG Trunk- Manager->QSIG ICT-> >QSIG Trunk->PBX Phone 1- >CFNA->QSIG Trunk-> >Unity UC712EF.QSG.005 QSIG per trunk Interaction with Callback Feature UC712EF.QSG.016 UC712EF.QSG.018 Call Diversion by Reroute Path Replacement in Trombone Call Call Diversion by Reroute,Extension Mobility and Voic Path Replacement Involving QSIG PBX and Unity Verify QSIG per trunk. Verify Call Diversion by reroute. Verify path replacement in Trombone call. SCCP Phone 1-> >QSIG ICT-> >QSIG ICT-> >QSIG trunk->westell->pbx Phone 1 Extension Mobility A-> Manager 1- >ICT- Manager 2-> IP Phone 1- >CFNA->ICT-> Manager 3- >Voic Phone 1-> Manager 1- >QSIG Trunk->PBX Phone 1- >XFER->QSIG Trunk-> Manager 1- >Unity 11 of 214

12 QSIG System Test Results for IP Telephony: Cisco System Release 7.1(3) UC712EF.QSG.020 Path replacement Verify path replacement in in Trombone Call Trombone call. Path Replacement in Trombone Call Involving DPNSS PBX and Two Line IP Phone Phone 1-> Manager 1- >ICT- Manager 2->Phone 1 (Line 1)- >Conference Bridge->Phone 1 (Line 2)->XFER->QSIG Trunk- >Westell->PBX Phone 1->XFER- >Westell->QSIG Trunk-> Manager 1- >Phone 2 12 of 214

13 Quality of Service System Test Results for IP Telephony: Cisco System Release 7.1(3) UCS713-QOS-010 QOS Wireless WAN QOS Testing Verify mixed voice (wired and wireless) and data traffic (Pageant) testing from remote site (T1 connected). Phone->Aironet Access Point- >LWAPP/WAN->Wireless LAN Controller-> Manager 13 of 214

14 UC Integration for MOC System Test Results for IP Telephony: Cisco System Release 7.1(3) UCS712IF.CSF.019 Personal Communicator Personal Communicator Client - Serviceability and Maintainability. Verify Serviceability and Maintainability of Personal Communicator Client on 1. Call Characteristics and Statistics, 2. Problem/Error reporting, 3. Server Health, 4. Debugging tool, 5. Logs keeping Verify Serviceability and Maintainability of Personal Communicator Client on 1. Call Characteristics and Statistics, 2. Problem/Error reporting, 3. Server Health, 4. Debugging tool, 5. Logs keeping. UC Integration for MOC Client1- Manager UCS712IF.CSF.018 UCS712IF.CSF.015 Personal Communicator Personal Communicator Personal Verify UC Integration for Communicator Client in Microsoft Office Communicator Phone Associated Mode client with Personal to IPv4/IPv6 Phone Communicator Plug-in installed Place an Emergency is associated with an IPV4/IPV6 Call phone. Place an Emergency call. Personal Communicator Client on Desk Phone Associated Mode to Extension Mobility IP Phone Picking Up a Group Pick Up DVO Call from Mobile Communicator Verify UC Integration for MOC with Personal Communicator Client Plug-in installed is associated to an Extension Mobility IP Phone desk phone. This client is part of a group pick up and group pick up an incoming DVO call set up by Mobile Communicator. UC Integration for Microsoft Office Communicator+CSF1- Manager->Emergency Responder UC Integration for MOC Client1- Manager1(Extension Mobility+Group Pickup) 14 of 214

15 UC Integration for MOC System Test Results for IP Telephony: Cisco System Release 7.1(3) UCS712IF.CSF.010 Personal Communicator UCS712IF.CSF.009 UCS712IF.CSF.007 Personal Communicator Personal Communicator Personal Communicator Client in Phone Associated Mode to a Secure IP Phone Set Up a Secure Ad-Hoc Conference Personal Communicator Client Set Up an Ad-Hoc Conference Involving Intercluster End Points Secure Personal Communicator Client in Transfer a Secure Call to a Secure End Point Verify if UC Integration for Microsoft Office Communicator Client with Personal Communicator Plug-in is phone associated mode to a secure IP Phone. Set up a secure conference. Verify if UC Integration for Microsoft Office Communicator Client with Personal Communicator Plug-in set up a 5 party ad-hoc conference involving UC Integration for Microsoft Office Communicator clients from another cluster, PSTN phones, and local end points. Verify if UC Integration for Microsoft Office Communicator Client with Personal Communicator Plug-in is in secure Softphone mode. A secure IP Phone calls this UC Integration for Microsoft Office Communicator client. Verify that secure indication is set on UC Integration for Microsoft Office Communicator client. Transfer the call to secure phone. UC Integration for Microsoft Office Communicator+CSF1- Manager->conference<-UC Integration for Microsoft Office Communicator+CSF2 UC Integration for Microsoft Office Communicator+CSF1- >Phone Proxy-> Manager 1- >Conference<- Manager 2<- UC Integration for Microsoft Office Communicator+CSF2<- PSTN IP Phone-> Manager 1- >UC Integration for Microsoft Office Communicator+CSF1- >Secure SIP-> Manager 2- >UC Integration for Microsoft Office Communicator+CSF2 15 of 214

16 UC Integration for MOC System Test Results for IP Telephony: Cisco System Release 7.1(3) UCS712IF.CSF.004 Personal Communicator Personal Communicator Client on Desk Phone Associated Mode to an IPv4/IPv6 IP Phone Answering the Call from MeetingPlace and Joining the Conference Call Verify if UC Integration for Microsoft Office Communicator Client with Personal Communicator Plug-in installed is associated to an IPV4/IPV6 IP desk phone. Establish a MeetingPlace conference and let MeetingPlace call the client. Answer the call, dial the Meeting password, and enter into the meeting. MeetingPlace-> Manager1->UC Integration for Microsoft Office Communicator+CSF UCS712IF.CSF.004 Personal Communicator Personal Verify UC Integration for Communicator Client on Desk Phone Associated Mode to an IPv4/IPv6 IP Phone Answering the Call From MeetingPlace and Joining the Conference Call Microsoft Office Communicator client with Personal Communicator Plug-in installed is associated to an IPV4/IPV6 IP desk phone. Establish a MeetingPlace conference and let MeetingPlace call the client. Answer the call, dial the Meeting password, and enter into the meeting. MeetingPlace-> Manager1->UC Integration for Microsoft Office Communicator+CSF UCS712IF.CSF.003 Personal Communicator Personal Communicator Client on Soft Phone Establishing a Secure Intercluster Call through Secure SIP Trunk Verify if UC Integration for Microsoft Office Communicator with Personal Communicator Plug-in installed is connected to the Manager through Phone Proxy. Establish an intercluster call to a secure phone over secure SIP trunk. UC Integration for Microsoft Office Communicator+CSF1- Manager1->Secure SIP-> Mobile Communicator2- >PhoneProxy-UC Integration for Microsoft Office Communicator+CSF2 16 of 214

17 UC Integration for MOC System Test Results for IP Telephony: Cisco System Release 7.1(3) UCS712IF.CSF.001 Personal Communicator Personal Communicator Client on Desk Phone Associated Mode to a IP Phone Establishing an Inter-cluster Call Across SIP Trunk Verify UC Integration for Microsoft Office Communicator client with Personal Communicator Plug-in installed is associated to a IP Phone desk phone. Establish an intercluster call to another UC Integration for Microsoft Office Communicator client connected through Phone Proxy across SIP trunk and check for bearer path. UC Integration for Microsoft Office Communicator+CSF1- Manager1-SIP-> Mobile Communicator2->PhoneProxy- UC Integration for Microsoft Office Communicator+CSF2 UC713EF.CSF.054 UC713EF.CSF.053 UC713EF.CSF.052 UC Integration for Microsoft Office Communicator UC Integration for Microsoft Office Communicator UC Integration for Microsoft Office Communicator UC Integration for Microsoft Office Communicator Interaction with IP Phone Conference with UC Integration for Microsoft Office Communicator and Normal Phones UC Integration for MOC Multi-party Conference Verify UC Integration for Microsoft Office Communicator client interaction with IP Phone. Verify Conference with UC Integration for Microsoft Office Communicator client and normal phones. Verify UC Integration for Microsoft Office Communicator multi-party conference. IP Phone-> Manager>UC Integration for Microsoft Office Communicator UC Integration for Microsoft Office Communicator-> Manager>SCCP Phone 1- >Conference->QSIG ICT- Manager>SIP Phone 1 UC Integration for Microsoft Office Communicator 1-> Manager>UC Integration for Microsoft Office Communicator 2->Conference- >UC Integration for Microsoft Office Communicator 3 17 of 214

18 UC Integration for MOC System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713EF.CSF.051 UC Integration for UC Integration for MOC Verify UC Integration for MOC 1 with UC Integration for Microsoft Office Communicator Click-to-call Feature Microsoft Office Communicator click-to-call feature. MOC- Manager->MOC 2 with UC Integration for MOC UC712EF.CSF.016 UC712EF.CSF.012 UC712EF.CSF.010 UC712EF.CSF.006 UC Integration for Microsoft Office Communicator UC Integration for Microsoft Office Communicator UC Integration for Microsoft Office Communicator UC Integration for Microsoft Office Communicator Microsoft Office Verify Microsoft Office Communicator (MOC) Communicator behavior when with UC Integration connection to Office Behavior when Server is lost. Connection to Office Server is Lost Microsoft Office Communicator (MOC) with Cisco UC interaction Microsoft Office Communicator Interaction with CCX Verify Microsoft Office Communicator with Cisco UC interaction Microsoft Office Communicator interaction with CCX as a caller phone. Microsoft Office Verify Microsoft Office Communicator with Communicator with Cisco UC Cisco UC Integration for Integration for Microsoft Office Microsoft Office Communicator interaction with Communicator Unity Voic . Interaction with Unity Voic Call from an Interoperability site SCCP phone to Microsoft Office Communicator with Cisco UC Integration for Microsoft Office Communicator Verify by making a call from an Interoperability site SCCP phone to Microsoft Office Communicator with Cisco UC Integration for Microsoft Office Communicator. MOC with UC Integration for MOC- Manager-> CCX->CAD Agent SCCP Phone-> >MOC with UC Integration for MOC->CFNA->Unity SCCP Phone 1-> Manager (Interoperability)->QSIG ICT- Manager->MOC with UC Integration for MOC 18 of 214

19 Performance System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713EL.PER.001 Load Load run with 12,000 BHCA in Medium Size Site Verify all kinds of call flows are executed for 5 days to achieve BHCA from a medium size site. SCCP Phone 1-> >ICT Trunk- Manager->SCCP Phone 2; SCCP Phone 1-> >Gatekeeper->IP-IP Gateway- >Gatekeeper-> CME- >SCCP Phone 2 UC713EL.PER.002 Load Load run with 36,000 Verify all kinds of call flows are BHCA in Large Size Site executed for 5 days to achieve BHCA from a medium size site. SCCP Phone 1-> >ICT Trunk- Manager->SCCP Phone 2; SCCP Phone 1-> >Gatekeeper->IP-IP Gateway- >Gatekeeper-> CME- >SCCP Phone 2 UC713EL.PER.003 Load Basic Load for SCCP / SIP calls with TLS encryption Verify all kinds of call flows are verified to/from phones registered to a cluster in secured mode. All phones are either secured or authenticated. SCCP Phone 1-> >MGCP Gateway-> >SCCP Phone 2; SIP Phone 1- Manager->MGCP Gateway- Manager->SIP Phone 2 19 of 214

20 Performance System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713EL.PER.004 Load Basic Load for SCCP calls Verify all kinds of call flows are verified by running the traffic for 24 hours. SCCP Phone 1-> >ICT Trunk- Manager->SCCP Phone 2; SCCP Phone 1-> >Gatekeeper->IP-IP Gateway- >Gatekeeper-> CME- >SCCP Phone 2 UC713EL.PER.005 Load Bulk Provisioning under Load Verify BAT tool by SCCP Phone 1-> adding/deleting/modifying large >ICT number of phones when calls are Trunk- being made to/from phones in Manager->SCCP Phone 2; parallel. SCCP Phone 1-> >Gatekeeper->IP-IP Gateway- >Gatekeeper-> CME- >SCCP Phone 2 UC713EL.PER.006 Load Upgrade under Load Verifies that Manager cluster is upgraded while the calls are continuously made to/from phones registered to Manager nodes. SCCP Phone 1-> >ICT Trunk- Manager->SCCP Phone 2; SCCP Phone 1-> >Gatekeeper->IP-IP Gateway- >Gatekeeper-> CME- >SCCP Phone 2 20 of 214

21 Attendant Console System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713EF.ARC.051 Attendant Console UC713EF.ARC.052 UC713EF.ARC.053 UC713EF.ARC.054 Attendant Console Attendant Console Attendant Console Consult Transfer between Two IP Phone 6900 Series Phones through Attendant Console Verify if operator can consulttransfer the call to the line 2 of IP Phone Blind Transfer Call Verify by making Blind Transfer Between IP Phone 6900 call between IP Phone 6900 Series Phone using Phone using Attendant Attendant Console. Console Attendant Console Call to a Deadline IP Phone 6900 Series Retrieval of Call Parked by Attendant Console Verify by making a call from IP Phone 6900 Phone 1 to Attendant Console to connect to another IP Phone 6900 Phone 2.This IP Phone 6900 Series Phone 2 line is down when Attendant Console is getting alerted. The Attendant Console should not connect the call to IP Phone 6900 phone 2. Verify retrieval of call parked by Attendant Console. Stage 1:IP Phone 6900 Phone 1- Manager-> Attendant Console (Operator Consoles Alerting) Stage 2: Attendant Console->Consult Transfer->IP Phone 6900 Phone 2(Line1) Stage 1:IP Phone 6900 Series Phone 1-> > Attendant Console(Operator Console Alerting) Stage 2: Attendant Console-> >Blind Transfer->IP Phone 6900 Series Phone 2. IP Phone 6900 Series Phone 1- Manager->Cisco Attendant Console->IP Phone 6900 Series Phone 2 IP Phone 6900 Series Phone 1- Manager> Attendant Console->Call Parking IP Phone 6900 Series; Phone 2->Park Retrieval 21 of 214

22 Attendant Console System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713EF.ARC.055 Attendant Failed CSCta64714 Console UC713EF.ARC.056 UC713EF.ARC.057 UC713EF.ARC.058 Attendant Console Attendant Console Attendant Console Presence Status of IP Phone 6900 Series Phones with Two Lines Multichannel Conferencing by Attendant Console Schedule MeetingPlace Express call to Attendant Console Accessibility Feature in Attendant Console Verify Presence status of IP Phone 6900 Series phones with two lines. Verify Multichannel conferencing by Attendant Console. Verify by scheduling MeetingPlace Express call to Attendant Console. Verify Accessibility feature in Attendant Console. Attendant Console- Manager->IP Phone 6900 (Line1) Stage 1: IP Phone 1- Manager-> Attendant Console; Stage 2: Attendant Console-> >Conference->Extension A; Stage 3: Attendant Console-> >Conference->Extension B Stage 1: MeetingPlace Express-> > Attendant Console Stage 2: Attendant Console->Blind Transfer- > IP Phone 1 Attendant Console- Manager> IP Phone 1 22 of 214

23 Attendant Console System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713EF.ARC.059 Attendant Console Answer Next on Queue Calls Based on Priority Verify Answer Next on Queue calls based on Priority. Stage 1: IP Phone 1- Manager-> Attendant Console->Queue1; Stage 2: IP Phone 1-> > Attendant Console- >Queue2; Stage 3: IP Phone 1-> > Attendant Console- >Queue3 UC713EF.ARC.060 Attendant Console Cherry Picking of Queue Calls on Attendant Console Verify Cherry Picking of Queue calls on Attendant Console. Stage 1: IP Phone 1- Manager-> Attendant Console->Queue1; Stage 2: IP Phone 1-> > Attendant Console- >Queue2; Stage 3: IP Phone 1-> > Attendant Console- >Queue3 23 of 214

24 Border Element System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713IF.CUB.006 ilbc Support with Verifies if XCODER is Invoked if MGCP Gateway the Codec Bit Rate < 256. XCODER is Invoked if the Codec Bit Rate < 256 Analog Phone->MGCP Gateway- Manager->SIP Trunk->IP-IP Gateway-> CME-> IP Phone UC713IF.CUB.005 UC713IF.CUB.004 UC713IF.CUB.003 Border Element Border Element Border Element isac Transcoding with Manager for Skinny Trunk to Voic System SIP EO <-> H.323 Fast Start Call With Repacketization srtp to RTP Interworking Verifies Transcoding support CME->SIP Trunkbetween isac, G.711, G.729, > Border Element->SIP G.722, and ilbc when Trunk- CME Phone deposits voic Manager->CFWD->Unity to HQ Subscriber. In this case due to Codec Mismatch between Manager and Unity region, Manager has to Invoke IOS Transcoder and CME phone should be able to leave voic . Verify by invoking Transcoder for different repacketization. Verify Interworking between srtp and RTP. Manager->G.729r8-> Border Element->G.729br8- > CME Manager->sRTP-> Border Element->RTP-> CME UC713IF.CUB.002 Border Element Border Element Verifies Border Element sends Invite to Active can forward call request to secondary Subscriber when primary is down by knowing Manager node When configured with OPTION Manager status via OPTION PING CLI PING CME-> Border Element-> Manager (Subscriber 1)->Subscriber 1 Goes Down-> Border Element sends INVITE to Manager (Subscriber 2) 24 of 214

25 Border Element System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713IF.CUB.001 Border Failed CSCta49687 Element isac Transcoding on Border Element for SIP / H.323 Trunk Verifies Transcoding support between isac, G.711. G.729, G.722, ilbc. UC701IF.CUB.007 UC701IF.CUB.003 G.722/iLBC Secure Transcoding and Conference Border Element DTMF Pass Through Secure Transcoding and Conference between G.722/iLBC and G.711 Manager Phone is Able to Leave Voice Mail to a CME Phone that is Call Forwarded to Unity Express via Border Element Verify that call between Manager and CME phone via IP-to-IP Gateway with securing transcoding and secure conference between Manager and CME participants. Verify Manager phone calls to SIP Phone registered with CME and CME phone call forward to Unity Express and Manager phone is able to leave voice mail. CM Phone->SIP Trunk- >IP-to-IP Gateway-> CME- > IP Phone->Conference- >IP-to-IP Gateway-> > IP Phone IP Phone-> >SIP trunk-> Border Element- >SIP Trunk-> Manager Express->SIP Phone->Unity Express UC700IF.CUB.008 ilbc Support with MGCP Gateway MGCP Call with ilbc Enabled Verifies call between analog phone plugged behind MGCP Gateway to CME Phone via IP-IP Gateway with codec set to ilbc under MGCP and codec set to ilbc between Manager and IP-IP Gateway Analog Phone->MGCP Gateway- Manager->SIP Trunk->IP-IP Gateway-> CME-> IP Phone 25 of 214

26 Border Element System Test Results for IP Telephony: Cisco System Release 7.1(3) UC700IF.CUB.004 Border Element DTMF pass through CME Phone able to Join Meeting Hosted by MeetingPlace via Border Element and Inbox Transcoder is Invoked for Codec Mismatch Verifies CME Phone calls MeetingPlace via SIP trunk and Joins the meeting via Border Element CME Phone->SIP Trunk- > Border Element->SIP Trunk- Manager->SIP Trunk-> MeetingPlace 26 of 214

27 CM Business Edition System Test Results for IP Telephony: Cisco System Release 7.1(3) UC702EF.SMB.004 Unity Connection Voice Mail Retrieval Verify the ability to successfully Stage 1:PSTN Phone 1->PSTN from Unity Connection retrieve a voice message in Unity Gateway-> on Manager Business Edition from SIP Phone Connection on Manager Business Edition from a SIP phone. Manager BE- >SCCP Phone 1->XFER->SIP Phone 1-> Manager BE- >Unity Connection Stage 2:SIP Phone 1-> Manager BE- >Unity Connection UC702EF.SMB.010 Unity Connection Voice Mail Deposit in Unity Connection Using G.729 Codec From Remote Site Verify the ability of depositing a voice message in Unity Connection using G.729 codec from a remote site. Stage 1:PSTN Phone 1->PSTN Gateway-> Manager BE- >Remote SCCP Phone 1->CFNA- Manager BE->Unity Connection Stage 2:Remote SCCP Phone 1- Manager BE->Unity Connection UC702EF.SMB.021 DND DND Feature During Callback Verify by making a call from a central SCCP phone which has Do Not Disturb (DND) feature activated to a remote SIP phone that is busy. The SCCP phone in the central site should have Callback activated. Stage 1:Remote SIP Phone 1- Manager Business Edition- >PSTN Gateway->PSTN Phone 1 Stage 2:SCCP Phone 1- Manager Business Edition- >Remote SIP Phone 1->Callback 27 of 214

28 CM Business Edition System Test Results for IP Telephony: Cisco System Release 7.1(3) UC702EF.SMB.025 Manager Commercial Hold Reversion with Extension Mobility Verify Hold Reversion with Extension Mobility. Create Extension Mobility profile that is similar to one of the SIP Phones Stage 1:PSTN Phone 1->PSTN Gateway-> Manager Business Edition->SIP Phone 1- Feature Intercom in the central site. Use this profile >Hold Stage 2:SIP Phone 1- to login to a remote phone. >Resume UC702EF.SMB.034 Manager Commercial Feature Intercom Intercom Between SIP and SCCP Phone on Different Remote Sites Verify by placing an intercom call between SIP phone on a remote site to a SCCP phone on another remote site while the SCCP phone on the first remote site is active on call. Remote SIP Phone 1->Intercom- Manager Business Edition- >Remote SCCP Phone 1 28 of 214

29 CM Express System Test Results for IP Telephony: Cisco System Release 7.1(3) SR60.CME Basic Call Flow CME-SCCP Verify by demonstrating the use IP Phone-> CME- Hardware Conference of hardware conference resource >WAN- to Manager within CME to Manager sites via Ad-Hoc conferencing Manager A-> CME- >AdHoc Conference->WAN- Manager B SR60.CME Basic Call Flow MeetMe Conference Verify by having a Meet-Me conference using hardware conference resources within CME. SR60.CME.010.2a Basic Call Flow VAD Conference Enable SR60.CME Basic Call Flow CME Conference Call Never Drops SR60.CME Basic Call Flow CME Support for Out of Dialog SR60.CME Basic Call Flow Release Transfer with Unity Connection Integrated to CME Verify by having VAD enabled or Disabled and perform conference calls across the WAN. Verify by demonstrating conference calls "never drop" when the conference initiator drops of the conference call. Verify to demonstrate the use of OOD-R from one CME site to remote CME site. Verify if a call from a Gatekeeper controlled Manager Express SCCP Phone through a IP-IP Gateway to Unity Connection integrated to Manager Express is successful and that Unity Connection is able to transfer the call. IP Phone 1-> CME 1->MeetMe Conference; CME 2->WAN->MeetMe Conference PC->Directory-> CME 1- > IP Phone-> CME 2-> IP Phone->Call Forward SIP Phone-> CME 1->SIP Trunk->IP-IP Gateway- >Gatekeeper-> CME 2- >Unity Connection->Release Transfer-> CME 2- > IP Phone. 29 of 214

30 CM Express System Test Results for IP Telephony: Cisco System Release 7.1(3) SR60.CME Basic Call Flow Call Between H.323 and SIP Site via IP-to-IP Gateway Involving a MGCP Gateway Verify if a call from a PSTN phone to an IP Phone registered to Manager Express can be call forwarded all to a remote Manager Express phone and be forwarded to Unity Express on no answer. PSTN Phone->MGCP Gateway- > CME 1->IP Phone- >CFA->Gatekeeper->IP-IP Gateway->SIP Trunk-> CME 2->IP Phone->CFNA- >Unity Express. SR60.CME Basic Call Flow Call from PSTN Phone Through H.323 Gateway Forwarded to Shared Line Verify if a call from a PSTN phone to an IP Phone registered to Manager Express can be forwarded to a remote Manager Express phone configured for shared line. PSTN Phone->H.323 Gateway- > CME 1-> IP Phone->CFA->Gatekeeper->IP- IP Gateway->SIP Trunk-> CME 2-> IP Phone (Shared line)->transfer-> > IP Phone. SR60.CME Basic Call Flow Hold and Resume Where the Call is Placed on Hold on SIP Network UC701IF.CME.014 CME Parallel Hunt without Transcoding on CME Verify if a call from a Gatekeeper controlled Cisco Manager Express SCCP Phone through a IP-IP Gateway to a Manager Express SIP Phone via SIP trunk can be placed on hold and resumed. Verify if parallel hunting works when the hunt member have SIP phone, SCCP phone and IP Phone SCCP Phone-> CME 1- >Gatekeeper->IP-IP Gateway- >SIP Trunk-> CME 2- >SIP Phone 30 of 214

31 CM Express System Test Results for IP Telephony: Cisco System Release 7.1(3) UC702EF.CME.005 CME H.450 Call Forward all from QSIG PBX on a CME to IPMA Manager Verify by making a call from a SIP proxy controlled SIP phone via Manager to QSIG PBX phone on Manager is call forward all to a QSIG PBX phone connected to CME which in turn has call forward busy to IPMA Manager. SIP Phone 1->SIP Proxy Server- >SIP- Manager->QSIG->PBX Phone 1- >Conference Bridge->QSIG- Manager->IP-IP Gateway (H.323)-> CME->QSIG- >PBX Phone 1->CFA->QSIG- > CME->IP-IP Gateway (H.323)-> >IPMA UC702EF.CME.015 Conference Ad-Hoc conference on CME with Verify by setting up Ad-Hoc conference by a local DPNSS PBX Phone and CME phone with a DPNSS PBX an IPMA Manager Phone Phone and an IPMA Manager Phone. Stage 1:SCCP Phone 1-> CME->IP-IP Gateway(H.323)- Manager->QSIG->Westell Gateway->DPNSS PBX Phone 1 Stage 2: SCCP Phone 1-> CME->CNF->CME->IP-IP Gateway(H.323)-> >IPMA Manager Stage 3:SCCP Phone 1-> CME->CNF- >DPNSS PBX Phone 1 & IPMA Manager 31 of 214

32 CM Express System Test Results for IP Telephony: Cisco System Release 7.1(3) UC702EF.CME.028 CME Dixie Cup HW conferencing Meet-Me Conference between QSIG PBX Phone on CME, a DPNSS PBX phone Verify by making a Meet-Me conference on CME between 2 CME phones ( SCCP and QSIG PBX phone connected to CME ) and a DPNSS PBX phone. Stage 1:SCCP Phone 1-> CME->CNF_MM Stage 2:PBX Phone 1->QSIG Trunk-> CME->CNF_MM Stage 3:DPNSS PBX Phone 1->Westell- >QSIG Trunk-> Manager>IP-IP Gateway (H.323)-> CME- >CNF_MM UC702EF.CME.078 Transfer to Voice Mail Transferring the Call from PSTN Phone to CME Phone Voice Mail using TrnsfVM Soft Key Verify by making a call from a PSTN phone (Phone A) to CME SCCP phone (Phone B) is forwarded on no answer to another CME SCCP phone (Phone C). The CME SCCP phone (Phone C) transfers the call to voice mail box of CME phone (Phone B) using TrnsfVM soft key. Stage 1:PSTN Phone 1->MGCP BRI- Manager->Gatekeeper->IP-IP Gateway->Gatekeeper-> CME->SCCP Phone 1->CFNA- > CME->SCCP Phone 2 Stage 2:SCCP Phone 2- >TrnsfVM->SCCP Phone 1 DN#- > CME->Unity Express(NM) Stage 3:SCCP Phone 1-> CME->Unity Express(NM) 32 of 214

33 CM Express System Test Results for IP Telephony: Cisco System Release 7.1(3) UC702EF.CME.156 Shared line cbarge and Privacy Shared Line cbarge and Privacy Support on CME with Consult Transfer to Manager Video Phone Verify by making a call from a QSIG PBX phone (Phone C) to SCCP CME shared line phone (Phone A) is barged from another CME shared line SCCP phone (Phone B). The call is consult transferred to a Manager video phone (Phone D) from CME phone (Phone A). Stage 1:PBX Phone 1->QSIG Trunk- Manager->Gatekeeper->IP-IP Gateway->Gatekeeper-> CME->SCCP Phone 1(SL) Stage 2:SCCP Phone 2 (SL)-> CME->cBarge->SCCP Phone 1 (SL)->XFER_C-> CME- >Gatekeeper->IP-IP Gateway- >Gatekeeper-> >SCCP Video Phone 1 UC713IF.CME.001 CME Parallel / Serial Hunt with Transcoder Verifies whether Transcoder is invoked on CME for Invoked on CME codec mismatch on IP for Codec Mismatch Phone 6921/6941/6961. UC713IL.CME.002 CME Calls Between IP Phone and CME over Inter-Cluster Trunk UC713IL.CME.002 CME IP to ICT to CME to IP Calls Verify that calls between IP Phone and CME over Inter-Cluster Trunk are successful. Verify that calls from SFO-ORD IP Phone to CME IP Phone over Inter-Cluster Trunk are successful. IP Phone 6921/6941/6961- Manager-> Border Element-> CME->Hunt Pilot->Parallel Hunt Group- >Xcoder Invoked->IP Phone 6921/6941/6961 UC713IL.CME.002 CME IP to ICT to CME to IP Calls Verify that calls from SFO-ORD IP Phone to CME IP Phone over Inter-Cluster Trunk are successful. 33 of 214

34 CM Express System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713L.CME.002 CME IP to ICT to CME to IP Calls Verify that calls from SFO-ORD IP Phone to Manager Express IP Phone over Inter-Cluster Trunk are successful. UC713L.CME.002 CME Calls from a IP Phone over ICT to CME IP Phone Verify that calls from SFO-ORD IP Phone to CME IP Phone over Inter-Cluster Trunk are successful 34 of 214

35 Manager System Test Results for IP Telephony: Cisco System Release 7.1(3) UCS713IF.CCM.014 Call Forward Use IP Phone 9951/9971 for PSTN Call Forwarded Across Verify that idivert feature can be invoked in ringing state. Also verify that the feature can be PSTN Phone->MGCP Gateway- Manager->IP Phone->Transfer- SIP Trunks and Divert used with calls forwarded across >SCCP IP Phone->CFA->SIP Feature Invoked During SIP trunks and the voic Trunk-> IP Phone->SIP Alerting State system integrated to Manager using SIP trunks and IP Phones. Trunk->Unity. UCS713IF.CCM.013 Basic Call Flow IP Phone 9951/9971 Using Directed Call Park Join Three Phones in to Conference Verify that an ad-hoc conference can be created using IP Phone 9951/9971 Join with a party who has already been directly parked and a new caller. Join 3 phones into conference using Directed Call Park IP Phones(3 Phones)- Manager->Directed Call park- >Conference Bridge UCS713IF.CCM.012 Basic Call Flow Device Mobility for IP Phone 9951/ Roaming User Adhoc Conference This will be filled in after more is learned about Device Mobility feature and functionality. IP Phone-> Manager Device Mobility->Conference UCS713IF.CCM.007 IP Phone 9951/9971 in a Conference Group Pick Up a Call and Chain the Conferences Two conferences. One of the party from conference 1 dial a number. One of the Party in conference 2 group pick the call and then chain the conferences. Conference 1-> IP Phone- >Chain Conference->Conference 2 UCS713IF.CCM.006 Park a Secure Call Using IP Phone 9951/9971 and Pick Up From another IP Phone Park an incoming SIP intercluster secure call. After the park reversion kick up, pick up the call from another secure IP Phone 9951/9971. Phone(Secure)-> Manager 1<SIP Trunk>-> Manager 2- > IP Phone 1->Call Park- > IP Phone 2 35 of 214

36 Manager System Test Results for IP Telephony: Cisco System Release 7.1(3) UCS713IF.CCM.005 cbarge and Set Up a Verify cbarge with 7900 series Conference Using Built In Conference Resource Involving IP Phones phones and IP Phones UCS713IF.CCM.004 IP Phone Use IP Phone to barge 9951/9971 Barge in to a into a secure call. Secure Call UCS713IF.CCM.003 IP Phone 9951/9971 Conference Using Built-in Conference Resource Verify by setting up a conference using built in resource of IP Phone 9951/9971. Use mix of authenticated and encrypted phones. Remove phones from conference and ensure call correctly switches between authenticated and encrypted modes. UCS713IF.CCM.002 Manager Secure Meet-Me Conference Using IP Phones 9951/9971 and TNP Phones UCS712IF.CCM.119 IPv6 Dual Stack SCCP Phone Behind SRST Gateway Verify by establishing a secure Meet-Me conference using IP Phones 9951/9971 and TNP phones. IP Phones->TNP Phone- Manager->Secure Meet-Me Conference Verify that dual stack phones can SCCP (v6/v4) (secure)-> fallback to SRST router when the Manager; WAN WAN link or connectivity to link Failure SCCP (v4) (Secure)- >SRST/SIP Gateway; Place a Manager is down. And also to call; WAN link restored SCCP verify that the dual stack phone (v6/v4) (secure)-> can register back to Manager Manager went it becomes reachable. 36 of 214

37 Manager System Test Results for IP Telephony: Cisco System Release 7.1(3) UCS712IF.CCM.118 IPv6 Early Offer Over Dual Stack SIP Trunk to a TRP Enabled Endpoint Verify that MTP is invoked when the endpoint has been configured with TRP. Also to verify that when the call is v6 then only one MTP is invoked for the call. SCCP (v6/v4)-> >SIP Trunk (v6/v4) (v6 media/sig pref) (ANAT on) (MTP)-> >MTP/TRP-> Personal Communicator UCS712IF.CCM.117 IPv6 Secure Conference Chaining of Two Conferences Involving Dual Stack Endpoints over Dual Stack SIP Trunk UCS712IF.CCM.115 IPv6 Out Dial from MeetingPlace to Dual Stack Phones across Dual Stack SIP Trunk and 3rd Party Video Endpoints Verify conference chaining behavior using the Join softkey with two conferences that use different Media Resource Gateway Lists (MRGLs). Verify that MeetingPlace can conference in dual stack phones across dual stack SIP Trunk and 3rd party video devices. SCCP (v4/v6) (secure) SIP Phone (secure); SCCP (v6/v4) (secure)->congbridge1; IP Communicator Personal Communicator PSTN Phone- >Conference Bridge2; Chain Conference SCCP (v6/v4)->conf- >IP Communicator APP Server->SIP-> Manager 1- >SCCP Video Advantage; APP Server->SIP- Manager 1->SIP Trunk-> Manager 2- >SCCP Video Advantage; APP Server->SIP- Manager 1->3rd Party video; APP Server->SIP-> Manager 1- > Personal Communicator 37 of 214

38 Manager System Test Results for IP Telephony: Cisco System Release 7.1(3) UCS712IF.CCM.114 IPv6 Call from a Dual Stack Verify that calls can be placed SCCP (v6/v4)-> Phone to Unity Express from a dual stack phone to CTI >CTI- AA Transferred to another Dual Stack Phone devices such as Unity Express. >Unity Express->AA-> >SCCP (v6/v4) UCS712IF.CCM.113 IPv6 ANAT enabled SIP Trunk to Unity Connection. UCS712IF.CCM.109 IPv6 Call Park over A SIP Trunk with Media and Signaling Preference Set To IPv6 UCS712IF.CCM.107 IPv6 Call Over a Dual Stack SIP Trunk Forwarded Over ICT to CME Verify if Unity Connection is v4 only application. This test case intends to verify if Unity Connection can answer a call where two m-lines (v4 and v6) are presented in the SDP of the INV. Verify by placing a call from a dual stack phone to a another dual stack phone where the SIP Trunk is also dual stack and is configured to prefer v6 for signaling and media. The call is then parked and retrieved by a 7925 phone. SCCP (v6/v4)-> >SIP Trunk (v6/v4) (ANAT-on) (MTP) (v6 sig and media pref)->unity Connection SCCP (v6/v4)-> >SIP Trunk (v4/v6) (v6 sig/media pref)- >SCCP (v6/v4)->call Park SCCP (v6/v4)-> >SIP Trunk (v4/v6) (v6 sig/media pref)- >7925 Verifies call placed from a dual SCCP Phone (v4/v6)-> stack phone to another dual >SIP stack phone where the SIP Trunk Trunk (v4/v6)-> has also been configured to Managersupport v4 and v6 but the media >SCCP Phone (v4/v6)- and signaling preference has >Conference Bridge->ICTbeen set to v6. The call is > CME->SIP Phone forwarded on busy to CME over ICT. 38 of 214

39 Manager System Test Results for IP Telephony: Cisco System Release 7.1(3) UCS712IF.CCM.107 IPv6 Call Over A Dual Stack SIP Trunk Forwarded Over ICT To CME UCS712IF.CCM.106 IPv6 PSTN Call To A Dual Stack VG224 And Call Answered By Group Pickup UCS712IF.CCM.105 IPv6 Secure Conferencing Over SIP Trunk Involving Dual Stack Phones and V4 Only Phone Verify by placing a call from a dual stack phone to another dual stack phone where the SIP Trunk has also been configured to support v4 and v6 but the media and signaling preference has been set to v6. The call is forwarded on busy to CME over ICT. Verify by placing a call from PSTN to IP Phone. The call is answered by a VG224 phone using Group PickUp. Group Pickup through VG224 is accessed via **4 on the FXS phone. Verifies secure conferencing over a SIP Trunk where the trunk has been configured for v6 media and signaling preference. SCCP Phone (v4/v6)-> >SIP Trunk (v4/v6)-> >SCCP Phone (v4/v6)- >Conference Bridge->ICT- > CME->SIP Phone PSTN Phone->SIP Gateway (v6/v4)->sip Trunk (v4 sig/v6 media pref)-> >VG224 Gateway (v6/v4)->ip Phone->GPickup->FXS phone. SCCP Phone 1-> >SIP- >SCCP Phone 2; SCCP Phone 2 (Conference)-> >Conference Bridge; SCCP Phone 1-> >SIP- Manager->MTP->Conference Bridge; SCCPPhone 2(v4) (encr)- Manager->Conference Bridge. 39 of 214

40 Manager System Test Results for IP Telephony: Cisco System Release 7.1(3) UCS712IF.CCM.103 IPv6 Video Call Over SIP Trunk with ANAT Enabled UCS712IF.CCM.102 IPv6 Music On Hold For A Call From PSTN Through Dual Stack SIP Trunk And SIP Gateway Verify by making a call from 7985 over a SIP Trunk with ANAT enabled to another 7985 phone. The call finally being transferred to a Personal Communicator with Video Advantage. Verifies a call from PSTN enters the cluster from a dual stack SIP Gateway and SIP Trunk. Test Music on Hold for the calling party (v4)-> >SIP Trunk (ANAT-on) (addressing mode v6/v4) (v6 sig pref) (v6 media pref)-> >7985 (v4)->xfer-> Personal Communicator PSTN->SIP Gateway->SIP Trunk- Manager->SCCP(v4)->Music on Hold- Manager->SIP Trunk->SIP Gateway->PSTN (Resume) PSTN->SIP Gateway->SIP Trunk- Manager->SCCP UCS712IF.CCM.101 IPv6 Blind Transfer of a IPV6 Call to CCX Phone Agent UC713L.CCM.120 Manager Verifies a call from a remote SCCP (v6/v4)-> cluster across a dual stack SIP >SIP trunk to dual stack SCCP phone. Trunk (ANAT-on) (addressing From the SCCP phone the call is mode v6/v4) (v6 sig pref/media blind transferred to CCX pref)- with a dual stack phone as agent. Manager->SCCP (v6/v4)->blind transfer-> CCX->SCCP Phone (v6/v4) IP Phone Agent Cluster Upgrade Collect upgrade time statistics. 40 of 214

41 Manager System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713L.CCM.108 Manager IP to IP Calls Verify that calls from AZO IP Phone to AZO IP Phone are successful. UC713L.CCM.020 UC713L.CCM.005 Manager Manager SFO-ORD Cluster Upgrade Calls between IP Phones Collect upgrade time statistics. Verify that calls from SFO-ORD IP Phone to SFO-ORD IP Phone are successful. UC713IL.CCM.120 Cluster Upgrade Collect upgrade time statistics. Manager UC713IL.CCM.120 AZO Cluster Upgrade Verify upgrades with IO throttling enabled such that while the system is under moderate load, no phones unregister, no code yellows occur, and the upgrade completes in a reasonable amount of time. UC713IL.CCM.108 UC713IL.CCM.100 Manager Manager Calls Between IP Phones Table Out of Sync Detection Service Parameter ON Verify that calls between IP Phones are successful. Table Out of Sync Detection service parameter is enabled to verify that no adverse affect of call processing occurs during DB table sync operations. 41 of 214

42 Manager System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713IL.CCM.020 Manager SFO-ORD Cluster Upgrade Verify upgrades with IO throttling enabled such that while the system is under moderate load, no phones unregister, no code yellows occur, and the upgrade completes in a reasonable amount of time. UC713IL.CCM.005 Manager Calls from IP Phone to IP Phone Verify that calls from one IP Phone to another IP Phone are successful. UC713IL.CCM.003 Manager IP to IP Video Calls Verify that calls from SFO-ORD IP Phone to SFO_ORD IP Phone are successful. UC713IL.CCM.001 Manager Enabling Table Out of Sync Detection Service Parameter Table Out of Sync Detection service parameter is enabled to verify that no adverse affect of call processing occurs during Database table sync operations. UC713IF.CCM011 Extension Mobility Mobile User Login to Extension Mobility of IP Phone and Activate Mobility-Mobile Connect and MVA Verify if Extension Mobility profile has Mobility Enabled and Remote destination set. Mobile user login to IP Phone 9951/9971. Verify that Mobile connect work on the phone. Switch the call between mobile phone and IP Phone 3 times. IP Phone-> Manager Extension Mobility-> Manager Mobility>PSTN->Mobile Phone 42 of 214

43 Manager System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713IF.CCM.022 Manager UC713IF.CCM.020 Manager Seamless Upgrade in Clustering over WAN Site and IP Phone 9951/9971 Resiliency when TFTP Server Restart and Manager Service Restart IP Phone DND UC713IF.CCM.010 Basic Call Flow Consult Transfer on Inter-cluster Calls Between Encrypted IP Phone 9951/9971 UC713IF.CCM.009 Manager Conference Chaining with IP Phone Using cbarge at Remote Site Verify the seamless upgrade and downgrade of IP Phone 9951/9971 can happen under moderate load and TFTP server restart or Manager service restart will not affect it's normal working. IP Phone-> Manager 1- >Clustering over WAN-> Manager 2- > IP Phone Verify if Do Not Disturb can be IP Phone 8961-> set at Manager Manager useradmin page as well as at IP Phone 8961 and behave as expected Verify consult transfer works on inter-cluster calls between encrypted IP Phone 9951/9971 across H.323 ICT. Verify conference chaining behavior using cbarge and when one of the conferences is at a remote site using conference bridge resource at Main campus. The test will also verify proper resource clearing when WAN is broken to the remote site. SIP Phone 1-> >ICT- Manager->SIP Phone 2->consult XFER>SIP Phone 3 Phone- Manager->Conference Bridge 43 of 214

44 Manager System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713IF.CCM.008 Manager Verify by ensuring that SIP trunks can support srtp on IP Phone. IP Phone Answer a Call to Hunt Pilot Over SIP Trunk with srtp Enabled. UC713IF.CCM.001 Shared Line Call Pickup and Blind Transfer using IP Phone 9951/9971 UC713EF.CCM.219 UC713EF.CCM.218 IM Session, Personal Communicator, Presence Video Calls and RSVP Inter Working in Remote Sites Personal Communicator Inter Working with Extension Mobility Video phones, IP Phones, Personal Communicator and Make Video Calls with RSVP IP Phone 1 in a hunt group. Incoming H.323 intercluster call to a hunt group number, IP Phones 2 picks up the call after Phone 1 rings then blind transfers the call to Phone 1 Verify Personal Communicator inter working with Extension mobility. Verify video phones, IP Phone 9951/9971, Personal Communicator by making video calls with RSVP. Phone A-> >SIP Trunk- Manager->Phone B; Phone A- Manager->SIP Trunk-> >Phone C IP Phone-> Manager 1- >H.323-> Manager 2- >Hunt Group-> IP Phone- >Blind XFER->Phone 2 UC713EF.CCM.217 IP Phone Interworking with Attendant Console Verify by calling from the remote IP Phone 9951/9971 to the Attendant Console which blind transfers the call to another IP Phone 9951/ of 214

45 Manager System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713EF.CCM.216 IP Phone Interworking with Unity Express Verify by making a call from the remote IP Phone 9951/9971 to the CME IP Phone 9951/9971 which has CFNA set to Voic . UC713EF.CCM.215 IP Phone Interworking with CCX UC713EF.CCM.214 IP Phone Interworking with MeetingPlace Express UC713EF.CCM.213 IP Phone Interworking with IP Communicator UC713EF.CCM.212 IP Phone Callback overrides DND using IP Phones 9951 and 9971 Verify by making a call from a PSTN phone to an SCCP phone is transferred via a QSIG ICT to CCX which transfers the call to a Cisco Agent Desktop (CAD) with a IP Phone 9951/9971. Verify by scheduling reservationless meeting with IP Phone 9951/9971 as one end point. Verify by making a call from the SIP IP Communicator to the Remote SCCP Phone over QSIG ICT which has CFNA set to a QSIG PBX Phone which has CFA set to a remote IP Phone 9951/9971. Verify Callback overrides DND using IP Phones 9951 and w/ Exception w/ Exception CSCsz89579 UC713EF.CCM.211 IP Phone Depositing a Voic Verify by depositing and Using IP Phones retrieving a voic using 9951, 9971 and Unity IP Phones 9951 and of 214

46 Manager System Test Results for IP Telephony: Cisco System Release 7.1(3) UC713EF.CCM.210 IP Phone Consult Transfer Using Verify consult transferring the call IP Phones 9951 to the second line of IP and 9971 Phone 9951/9971 when the first line is busy. UC713EF.CCM.209 IP Phone Call Park and Park Retrieval using IP Phones 9951 and 9971 UC713EF.CCM.208 IP Phone Hold Reversion in IP Phones 9951 and 9971 UC713EF.CCM.207 IP Phone 6900 Series UC713EF.CCM.206 IP Phone 6900 Series + Dialing in IP Phone 6900 Series Phones in SRST Mode IP Phone 6900 Series Phones in SRST Mode Verify Call park and park retrieval using the IP Phones 9951/9971. Verify Hold Reversion Timer expiry in IP Phones 9951 and Verify + Dialing in IP Phone 6900 Series Phones in SRST mode. Verify IP Phone 6900 Series Phones in SRST mode. w/ Exception UC713EF.CCM.205 IP Phone 6900 Series IP Phone 6900 Series Phones in SRST Mode Verify IP Phone 6900 Series Phones in SRST mode. UC713EF.CCM.204 IP Phone 6900 Series Depositing a VM Using IP Phone 6900 Series Phone Using Unity Connection in dual stack cluster Verify by depositing a VM from a IP Phone 6900 Series Phone to a VG224 DS Phone over H225 Trunk to a non dual stack cluster and then over a QSIG ICT to a dual stack cluster. w/ Exception 46 of 214