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KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 1 of 30 Polycom8440CS1K Avaya Solution & Interoperability Test Lab Application N otes for Polycom SpectraL ink 8440 SIP Telephone version 4.0.0. 0282 with Avaya Communication S erver 1000 R elease 7.5 – Issue 1 . 1 Abstract These Application Notes describe a solution comprised of Avaya Communication Server 1000 SIP Line Release 7.5 and Polycom SpectraLink 8440 SIP telephone . During the compliance testing, the Polycom SpectraLink 8440 was able to register as a SIP c lient endpoint with the Communication Server 1000 SIP Line gateway . The Polycom SpectraLink 8440 telephone was able to place and receive calls from the Communication Server 1000 Release 7.5 non - SIP and SIP Line clients. The c ompliance tests focused on basic telephone features. Information in these Application Notes has been obtained through DevConnect compliance testing and additional technical discussions. Testing was conducted via the DevConnect Program at the Avaya Soluti on and Interoperability Test Lab. KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 2 of 30 Polycom8440CS1K 1. Introduction These application notes provide detail configurations of Avaya Communication Server 1000 SIP Line release 7.5 (hereafter refer red to as CS 1000) and the Polycom SpectraLink 8440 SIP telephone Version 4.0.0. 0282 u sed during the comp liance testing. The Polycom SpectraLink 8440 was tested with non - SIP and SIP clients using the CS1000 SIP line release 7.5 . All the applicable telephony feature test cases of release 7.5 SIP line were executed on the Polycom SpectraLink 8440 , where applicable, to ensure that the interoperability with CS 1000. 2. General Test Approach and Test Results The general test approach was to have the Polycom SpectraLink 8440 telephone to register to the CS1000 SIP line gateway successf ully. From the CS1000 telephone client s /user s to place a call to and from the Polycom SpectraLink 8440 telephone and to exercise other telephony features such as busy, hold, DTMF , MWI and codec negotiation 2.1. Interoperability Compliance Testing The focus of this testing was to verify that the Polycom SpectraLink 8440 SIP telephone was able to interoperate with the CS 1000 SIP line system. The following areas were tested:  Registration of the Polycom SpectraLink 8440 SIP telephone to the CS1000 SI P Line Gateway.  Call establishment of Polycom SpectraLink 8440 with CS1000 SIP and non - SIP tele phones.  Telephony fea tures: Basic calls, conference , transfer, DTMF ( dual tone multi frequency) RFC2833 and INBAND transmission, voicemail with Message Waiting Indication (MWI) notification, busy, hold, speed dial, group call pickup, call waiting, ring again busy/no answer, multiple appearances Directory Number.  PSTN calls over PRI trunk.  Codec negotiation – G . 711 and G . 729. 2.2. Test Results The objectives outline d in the Section 2.1 were verified . The following observations were made during the compliance testing :  Avaya has not performed audio performance testing or reviewed the Polycom SpectraLink 8440 compliance to required industry standards .  Polycom SpectraLink 8440 does not support DTMF via S IP INFO . DTMF default s as INBAND . When using RFC2833 DTMF set these fields tone.dtmf.rfc2833Control ="1" and tone.dtmf.rfc2833Payload=101 in the config file (sip.cfg) .  enabled but it will be not used for the busy call since when the 8440 phone is in busy status the Server Call Forward Busy feature of CS1000 SIPLine will take place before it KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 3 of 30 Polycom8440CS1K can be executed by the phone. 1000 SIPLine server.  It is highly recommended to disable the media security on the Call Server to avoid some unexpected behaviors such as one way audio from a call m ade from PSTN over a PRI trunk.  There is a limitation for the l ocal conference on the Polycom 8440 phone with Avaya UNIStim phone . When Polycom 8440 acts as the moderator of the conference and conference s with two Avaya UNIStim phones , the conference is su ccessfully opened with 3 - way audio. However, after the Polycom 8440 phone disconnects, the two Avaya UNIStim phones remain c onnected but have no speech path in between them . This issue doesn’t happen with Avaya SIP 1140 phone s and with other SIP phone s . 2.3. Support For technical supp ort for the Polycom SpectraLink 8440 SIP endpoints , please contact Polycom Inc technical support as shown below: 1.800.POLYCOM or +1.925.924.6000 www.polycom.com 3. Reference Configuration Figure 1 illustrates the test configuration used during the compliance testing between the Avaya CS1000 and the Polycom SpectraLink 8440 . KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 4 of 30 Polycom8440CS1K Figure 1 : Network Configuration Diagram 4. Equipment and Software Validated The following equipment and software was used during the lab testing: Equipment Software Version Avaya CS1000 E Call Server (CPPM): 7.50Q Signaling Server (CPPM): 7.5 0. 17 Avaya CallPilot™ Messaging System 5.0.1 Avaya IP Soft Phone 2050 Avaya IP Phone 1140 Avaya IP Phone 2004P2 Avaya IP Phone 2002P2 3.04.0003 0625C6O 0692D93 0604DC5 Avaya SIP 1140 02.02.21.00 Polycom SpectraLink 8440 4.0.0. 0282 Polycom SpectraLink 8450 4.0.0. 0282 Provisioning Server OS Windows Vista x86 5. Configure Avaya CS 1000 - SIP LINE This section describes the steps to configure the Avaya CS1000 SIP Line using CS 1000 Element Manager. A command line interface (CLI) option is available to provision the SIP Line application on the CS 1000 sy stem. For d etailed information on how to configure and administer the CS 1000 SIP Line , please refer to the S ection 9 [1] . The following is the summary of tasks needs to be done for configuring the CS 1000 SIP Line : - Log in to Unified Communications Management ( UCM) and Element Manager (EM). - Enable SIP Line Service and Configure the Root Domain. - Create SIP Li ne Telephony Node. - Create D - Channel for SIP Line . - Create an Application Module Link (AML). - Create a Value Added Server (VAS). - Create a Virtual Trunk Zone. - Create a Rou te Data Block (RDB). - Cr eate SIP Line Virtual Trunks. - Creat e SIP Line phones. 5.1. Prerequisite This document assumes that the CS1000 SIP Line server has been: - Installed with CS 1000 Release 7.5 Linux Base. KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 5 of 30 Polycom8440CS1K - Joined CS 1000 Release 7.5 Security Domain. - Deployed with SIP Line Application. The f ollowing packages need to be enabled in the key code . If any of these features have not been enabled, please contact your Avaya account team or Avaya technical support at http://w ww.avaya.com . Package Mnemonic Package # Descriptions Package Type Applicable market SIP_LINES 417 SIP Line Service package New package Global FFC 139 Flexible Feature Codes Existing package Global SIPL_ AVAYA 415 Avaya SIP Line package Existing package Global SIPL_3RDPARTY 416 Third - Party SIP Line Package Existing package Global 5.2. Log in to Unified Communications Management (UCM) and Element Manager (EM) Us e the Microsoft I nternet Explorer browser to launch CS 1000 UCM web portal at http://IP Address or FQDN where IP address or FQDN&#x-5I1;P-3;&#x a4d; r-6;äss;&#x or ; &#xQD4N; is the UCM Framework IP address or FQDN for UCM server. Log in with the username/password which was defined during the primary security ser ver configuration, the UCM home page appears as shown in the Figure 2 be low. KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 6 of 30 Polycom8440CS1K Figure 2 : The UCM Home Page of CS 1000 Release 7.5 On the U CM home page, under the Element Name column, click on the EM name of CS 1000 system that needs to be configured, in t his sample that is cpp p m3 . The CS 1000 Element Manager page appears as show n in Figure 3 below. KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 7 of 30 Polycom8440CS1K Figure 3 : CS 1000 Release 7.5 EM Home Page 5.3. Enable SIP Line Service in the Customer Data Block On the EM page, navigate to Customers on the left column menu; select the customer number to be enabled with SIP Line Service (not shown). - Enable SIP Line Service by clicking on the SIP Line Service check box. - Enter the prefix number in the User agent DN prefix text box as shown in Figure 4 . Figure 4 : SIP Line Service in Customers Data Block KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 8 of 30 Polycom8440CS1K 5.4. Add a new SIP L ine Telephony Node On the EM page, navigate to menu System  IP Network  Nodes: Servers, Media Cards . Click Add to a dd a new SIP Line Node to the IP Telephony Nodes. The new IP Telephony Node page appears as shown in Figure 5 . Enter the information as show n below : - Node ID text box : 512 - this is the node ID of SIP Line server. - Call Server IP Address text box: 10 .10.97.78. - Node IP Address text box: 10 .10.97.187 - this is the IP address that SIP endpoint uses to register to. - Subnet Mask text box: 255.255.255.192. - Embedded LAN (ELAN) Gateway IP Address text box: 10 .10.97.66. - Embedded LAN (ELAN) Subnet Mask text box: 255.255.255.192. - Check SIP Line check box to enabl e SIP Line for this Node . Figure 5 : Adding a New IP Telephony Node - Click on the Nex t button to go to next page . The page, New IP Telephony Nod e with Node ID, will appear as shown in Figure 6 . - On the Select to Add drop down menu list, select the desired server to add to the node. - Click the Add button - Select the check box next to the newly added server, and click Make Leader (not shown). KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 9 of 30 Polycom8440CS1K Figure 6 : Adding a New IP Telephony Node (cont) - Click on the Nex t button to go to next page . The SIP Line Configuration Detail page appears as shown in Figure 7 . - Enter SIP Line domain name in SIP Domain name t ext box, for example sipl75.com . Figure 7 : Adding a new IP Telephony Node (cont) KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 10 of 3 0 Polycom8440CS1K - Under the SIP Line Gateway Services section, select MO from the SLG Role list. - From the SLG Mode list, select S1/S2 (SIP Proxy Server 1 and Server 2) , see Figure 8 . Figure 8 : Adding a new IP Telephony Node (cont) - Click Next . The page appears (not shown). - Click on the Transfer Now button and then T he Synchronize Configuration Files (Node ID 512 ) page appears. - Click Finish and wait for the configuration to be saved. The Node Saved page appears , see Figure 9 . KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 11 of 30 Polycom8440CS1K Figure 9 : Node Saved with Transfer Configuration - S elect the SIP Line server that associated with changes and then click on the Start Sync button to transfer the configuration files to the selected servers , see Figure 10 . Figure 10 : Synchronize Configuration Files Note : The first time a new Telephony Node is added and transfer ed to the call server, the SIP Line services need to be restarted. To restart the SIP Line services, log in a s administrator to the command line interface of the SIP Line server and issue the command: appstart restart . KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 12 of 30 Polycom8440CS1K 5.5. Create a D - Channel for SIP Line On the EM page, on the left column menu navigate to Routes and Trunks - D - Channels . Under the Configuration section as shown in Figure 11 , enter a number in the Choose a D - Channel Number field , and click on the to Add button . Figure 11: D - Channels configuration page - The D - Channels xx Property Configuration page appears as shown in Figure 1 2 . - From the Interface type for D - channel (IFC) list, select Meridian Meridian1 (SL1) . - Leave the other fields at default values. KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 13 of 30 Polycom8440CS1K Figure 12: SIP Line D - Channel Property Configuration - Click on the Basic options (BSCOPT) link. The Basic options (BSCOPT) list expands (not shown). - Click on Edit to configure Remote Capabilities (RCAP) . The Remote Capabilities Configuration detail page will appear as shown in Figure 1 3 . - Select the Message waiting interworking with DMS - 100 (MWI ) check box. - Select the Network name display method 2 (ND2) check box. - At the bottom of the Remote Capabilities Configuration page, click Return - Remote Capabilities to return the D - Channel xx Property Configuration page . KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 14 of 30 Polycom8440CS1K Figure 13: SIP Line D - Channel RCAP Configuration Details - Me ssage Waiting Interworking with DMS - 100 (MWI) must be enabled to support voice mail notification on SIP Line endpoints. - Network Name Display Method 2 (ND2) must be enabled to support name display between SIP Line endpoints . - Other check boxes are left unchecked. Click on the Submit button of the D - Channel Property Configuration page to save changes. 5.6. Create an Application Module Link (AML) On the EM page, navigate to System - Interfaces - Application Module Link , c lick on the Add button to add a new Application Module Link (not shown) . The New Application Module Link page appears as shown in Figure 1 4 . Enter an AML port number in the Port number text box. The AML of SIP Line Service can use a port from 32 to 127. In this case, SIP Line Serv ice is con figured to use port 3 3 . Click on the Save button to complete adding the AML link , and to save the configuration. KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 15 of 30 Polycom8440CS1K Figure 14 : Adding a new AML 5.7. Create a Value Added Server (VAS) On the EM page, navigate to System - Interfaces - Value Added Server and c lick on the Add button to add a new VAS . The Value Added Server page appears (not shown) , in this page, select the Ethernet Link link and the Ethernet Link page appears as shown in Figure 1 5 . Enter a number in the Value added server ID field, in this example 33 was used . In the Ethernet LAN Link drop down list, select the AML number of ELAN that was created in the S ection 5 .6 . Leave other fields as default values and click on the Save VAS and save the configuration. KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 16 of 30 Polycom8440CS1K Figure 15: Adding a new Value Added Service for the AML 5.8. Create a Virtual Trunk Zone On the EM page, navigate to menu System - IP Network - Zones . The Zones page appears on the right , in this page select Bandwidth Zones link . On the Bandwidth Zones page, click on the Add button, the Zone Basic Property and Bandwidth Management page appears as shown in Figure 16 . Enter a zone number in the Zone Number (Zone) field and in the Zone Intent (ZBRN) drop down menu select VTRK (VTRK) . L eave other fields as default values and click on the Save the Zone. Note : Repeat the step above to create another zone for the SIP Line phone; however remember to select MO , instead of VTRK in the field Zone Intent . KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 17 of 30 Polycom8440CS1K Figure 16 : Adding a new Zone for Virtual Trunk 5.9. Create a SIP Line Rout e Data Block (RDB) On the EM page, navigate to the menu Routes and Trunks - Routes and Trunks ; the Route s and Trunks page appears (not shown). In this page, click on the Add route button ne xt to the customer number that the route will belong to. The Customer ID, New Route Configuration page appears, expand the Basic Configuration tab, and enter values below and as shown in Figure 1 7 and 1 8 . - Route Number (ROUT) : 3 - Trunk t ype (TKTP) : TIE - Incoming and Out go ing trunk (ICOG) : IAO - Access Code for Trunk group (ACOD) : enter a number for ACOD, for example 757. - The route is f or a virtual trunk route (VTRK) : Checked. - Zone for codec selection and bandwidth management (ZONE ) : 4, this is the Virtual trunk zon e number that created in the S ection 4.8 . - Node ID of signaling server of this route (NODE) : 512, this is t he node ID of the SIP Line . - Protocol ID for the route (PCID) : SIP Line (SIPL). - : checked . - Mode of operation (MODE) : Route uses ISDN Signaling Link (ISLD) . - D channel number (DCH) : 4 , the D - channel number that was created in the Section 4.5 . - Interface type for route (IFC) : Meridian M1 (SL1). - Network calling name allowed (NCNA) : checked. - Channel type (CHT P) : B - channel (BCH). - Call type for ou tgoing direct dialed TIE route (CTYP ) : CDP. - Calling Number dialing plan (CNDP) : CDP. KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 18 of 30 Polycom8440CS1K Leave default values for The Basic Route Options, Network Options, General Options, and Advanced Configurations sections. Click the Submit button to complete adding the route and save configuration. Figure 11 7 : SIP Line Route Configuration KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 19 of 30 Polycom8440CS1K Figure 1 8 : SIP Line Route Configuration (cont) 5.10. Create SIP Line Virtual Trunk s On the EM page, navigate to Routes and Trunks - Routes and Trunks and select the Add route button beside to the route was created in the Section 5 .9 above to create new trunks. The Customer ID, Route ID, and Trunk type TIE trunk data block page appears as shown in Figure 1 9 , enter values for field s as shown below: - Multiple trunk input number (MTINPUT) : 32 - create 32 trunks. - Auto increment member number : checked. - Trunk data block (TYPE) : IP Trunk (IPTI). - Terminal Number (TN) : 100 0 2 0 - enter the first TN of a range TN. - Member number : 33, this is ID of trunk, just enter th e first ID for first trunk, next ID will be automatically created and incremented. - Start arrangement Incoming : Immediate (IMM). - Start arrangement Outgoing : Immediate (IMM). - Trunk Group Access Restriction (TGAR) : 1. - Cha nnel ID for this trunk : 33, this ID should be the same with the ID of Member Number. Click on the Class of Service button and assign following class of services (not shown) : - Media security : Media Security Never (MSNV). - Restriction level : Unrestricted. KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 20 of 30 Polycom8440CS1K Leave other fields a t default values and click on the Return Class of Service button to return to the Trunk type TIE trunk data block page . Click on the Save Figure 1 9 : Adding virtual trunks for SIP Line Trunk 5.11. Create a SIP Line Phone To create a SIP Line phone on the Call Server, log in as administrator using the command line interface (CLI) and issue the overlay (LD) 11 /20 as shown below. The bold fields must be properly inputted as they ar e co nfigured on the Call server, for other fields hit enter to leave it a t default values. KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 21 of 30 Polycom8440CS1K KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 22 of 30 Polycom8440CS1K 6. Configure Polycom SpectraLink 8440 This section describe s how to access the Polycom SpectraLink 8440 SIP endpoint web interface and configure the Polycom 8440 for testing. For more information on how to configure the Polycom SpectraLink 8440 phone connected to the Access Point Wi - Fi router, please refer to the document in the Section 9 . 6.1. Login Polycom SpectraLink 8440 This section shows how to log in to the home page of Polycom SpectraLink 8440 to manage and configure the 8440 phone . Open the web browser, and in the address field enter the Polycom SpectraLink 8440 IP address : http://ipaddress and t he Polycom SpectraLink 8440 login page will appear as shown in Figure 20 . Enter the username , Polycom , and its default password , 456 . Figure 20 : Polycom SpectraLink 8440 Login Screen Click the Submit button, t he homepage of Polycom SpectraLink 8440 appears as in Figure 2 1 below. KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 23 of 30 Polycom8440CS1K Figure 2 1 : Home page of Polycom SpectraLink 8440 telephone 6.2. Configure the Lines for Polycom SpectraLink 8440 This section shows how to configure the Polcom 8440 telephone to register with the CS1000 SIP Line gateway . On the homepage of the configuration screen (see Figure 2 1 ), click on the me nu, the Si page appears as shown in Figure 2 2 . Enter the values as shown below: - Language : select English (Internal) in the Phone Language drop down menu. - Time Synchronization : sel ect time zone for phone, for example (GMT - 5) Eastern Time (US and Canada) . - SIP Server: o Address : 1 0 .10.97.187 - t his is IP address of CS 1000 SIP Line server. o Port: 5070 - SIP Outbound proxy : o Address : 1 0 .10.97.187 - Use the same IP as the CS 1000 SIP Line server. o Port : 5070 - SIP Line Identification : o Display Name : Poly 8440 KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 24 of 30 Polycom8440CS1K o Address : 54008@sipl75.com o Authentication User ID : 54008 - this user ID was configured in the field SIPU when creating TN of SIP Line phone in the Section 5 .11 o Authentication Password : 1234 - this password was configured in the field SCPW when creating TN for SIP Line phone in the Section 5 .1 o Label : Click on the Save button to save changes. Figure 2 2 : KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 25 of 30 Polycom8440CS1K 6.3. L “Local Call Forward” such as Call Forward All call s , Call forward busy and Call Forward No Answer on the Polycom SpectraLink 8440 telephone. On the homepage of Polycom 8440 (see Figure 2 1 ) , navigate to menu Setting - Lines - Call Diversion , the Call Diversion section appears as shown in Figure 2 3 . - To set the Forward All Call s , select the Enable option button in the line Forward All . - To set the Forward Busy, select the Enable option butt on in the line On Busy and enter a forward DN on the Busy Contac t box. - To set the Call Forward No Answer, select the Enable option button in the line On No Answer. Note : The “ Server Call Forward Always ” setting for the Polycom 8440 on the CS 1000 Call Server must be OFF in order to make the “ Local Call Forward Always ” on the Polycom 8440 take e ffect. Figure 2 3 : Call Diversion section of Polycom 8440 KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 26 of 30 Polycom8440CS1K 6.4. Codec This section s the Codec on the Polycom SpectraLink 8440 phone. The compliance t esti ng has been done on both codecs, G711 and G729. On the homepage of Polycom SpectraLink 8440 (see Figure 2 1 ), navigate to menu Preference - Codec Preferences , the Audio and Video Codec Preferences page appears as shown in Figure 2 4 . The list of audio Codec s that are being used appear under the In use column. To use the codec G711 as the first choice, move it up to the top of the In Use list, repeat the same for other codecs if it needs to be the f irst choice. Click on the Save button to save changes. Figure 2 4 : Audi o & Video Codec Preferences KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 27 of 30 Polycom8440CS1K 7. Verification Steps This section includes some steps that can be followed to verify the configuration.  Verify that the Poycom SpectraLink 8440 telephone registers successfully with the CS 1000 SIP Line Gateway server and Call Server by using the CS 1000 Linux command line and CS 1000 Call Server overlay LD 32. - Log in to the SIP Line server as an administrator by using Avaya account. - Issue comman d “ slgSetShowByUID [userID]” where userID is SIP Line user’s ID being checked. [ admi === VTRK === UserID AuthId TN Clients Calls --------------- ---------- ---- ----------- ------- ----- --- ------ ------- --------- 54008 54008 104 - 00 - 00 - 01 1 0 0x8fc4cf8 SIP Lines StatusFlags = Registered Controlled KeyMapDwld SSD FeatureMask = CallProcStatus = 0 Current Client = 0, Total Clients = 1 == Client 0 == IPv4:Port:Trans = 10 .10.98.55:5060:udp Type = SIP3 UserAgent = PolycomSpectraLink - SL_8440 - x - nt - guid = 267d228547c1562399f1f743a2971fb5 RegDescrip = RegStatus = 1 PbxReason = OK SipCode = 200 hTransc = (nil) Expire = 3600 Nonce = f56 a9946ba497bde7eb445efb518f4f1 NonceCount = 2 hTimer = 0x8f64e60 TimeRemain = 1338 Stale = 0 Outbound = 0 ClientGUID = 0 MSec CLS = MSNV (MSEC - Never) C ontact = sip:54008@135.10.98.55:5060 KeyNum = 255 AutoAnswer = NO Key Func Lamp Label 0 3 0 54008 1 126 0 2654008 2 9 0 3 29 0 4 22 0 5 2 0 54334 KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 28 of 30 Polycom8440CS1K 17 16 0 18 18 0 19 27 0 20 19 0 21 52 0 22 25 0 24 11 0 25 30 0 26 31 0 == S ubscription Info == Subscription Event = None Subscription Handle = (nil) SubscribeFlag = 0 - Log in to the call server using the admin account. - Load overlay 32 and then issue command “stat [TN]” where TN is the SIP Line user’s T N being checked ld 32 NPR000 .stat 104 0 0 1 IDLE REGISTERED 00  Place a call from and to Polycom SpectraLink 8440 telephone and verify that the call is est ablished with 2 - way speech path.  During the call , use a pcap tool (ethereal/wireshark) at the SIP Line Gateway and clients to make sure that all SIP request/response messages are correct. 8. Conclusion All of the executed test cases have passed and met the objectives outlined in the Section 2 .1 , with some exceptions outlined in S ection 2 .2 . The Polycom S pectraLink 8440 version 4.0.0. 0282 is considered to be in complian ce with Avaya CS 1000 SIP Line System Release 7.5 . 9. Additional References Product documentation for the Avaya CS 1000 products may be found at: https://support.avaya.com/css/Products/ Product documentation for the Polycom SpectraLink 8400 series products may be found at: http://www.polycom.com [1] Avaya CS10 00 Documents: Avaya Communication Server 1000E Installation and Commissioning Avaya Communication Server 1000 SIP Line Fundamental, Release 7.5 Avaya C ommunication S erver 1000 Element Manager System Reference – Adm inistration KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 29 of 30 Polycom8440CS1K Avaya C ommunication S ever 1000 Co - resident Call Server and Signaling Server F undamentals Avaya Communication Server 1000 Unified Communications Management Common Services Fundamentals . Avaya Communication Server 1000 ISDN Primary Rate Interface Installation and Commissioning [2] Polycom SpectraLink 8400 Series Documents: Administrator’s Guide for the Polycom® UC Software Polycom® SpectraLink® 8400 Series Wireless Telephone Deployment Guide KP; Reviewed: SPOC 6 / 20 /2011 Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved. 30 of 30 Polycom8440CS1K © 201 1 Avaya Inc. All Rights Reserved. Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and ™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the property of their respective owners. The information provided in these Application Notes is subject to change without notice. The configurations, technical data, and recommendations provided in these Application Notes are believe d to be accurate and dependable, but are presented without express or implied warranty. Users are responsible for their application of any products specified in these Application Notes. Please e - mail any questions or comments pertaining to these Applicat ion Notes along with the full title name and filename, located in the lower right corner, directly to the Avaya DevConnect Program at devconnect@avaya.com .