Configuring the IP Trade System with Avaya SIP Enablement Services and Avaya Communication Manager - Issue 1.0
Abstract
These Application Notes describe the procedures for configuring the IP Trade system to communicate via a SIP interface with Avaya SIP Enablement Services and Avaya Communication Manager. The IP Trade system is an IP based trading floor solution. It consists of a set of IP Trade Profile-IK turrets, a Turret Support Server (TSS), and a Turret Proxy to Open Line Dealing Server (TPO). The IP Trade trading turrets register as SIP endpoints with Avaya SIP Enablement Services. This enables the IP Trade Profile-IK turret to integrate with Avaya Communication Manager and Avaya SIP Enablement Services. Information in these Application Notes has been obtained through interoperability compliance testing and additional technical discussions. Testing was conducted via the DeveloperConnection Program at the Avaya Solution and Interoperability Test Lab.
1. Introduction These Application Notes describe the procedures for configuring the IP Trade system to communicate via a SIP interface with Avaya SIP Enablement Services and Avaya Communication Manager. The IP Trade system is an IP based trading floor solution. It consists of a set of IP Trade Profile-IK turrets, a Turret Support Server (TSS), and a Turret Proxy to Open Line Dealing Server (TPO). The IP Trade trading turrets register as SIP endpoints with Avaya SIP Enablement Services. This enables the IP Trade Profile-IK turret to integrate with Avaya Communication Manager and Avaya SIP Enablement Services.
• TSS server: It provides security extensions, end user profiles management, hunt group, and bridge to middle-office applications.
• IP Trade Profile IK turret: The IP Trade Profile-IK turret is SIP-based VoIP trading phone. Its call handling panel uses Avaya Communication Manager feature name extensions (FNE) and/or IP Trade Profile internal features to provide the following call features:
o Outgoing calls o Incoming calls o Place a call on hold o Transferring a call o Parking a call o Forwarding a call o Call logging o Call monitoring o Conferencing (Three party conference) o Voice mail (Message waiting indication) o Call dispatcher o Barge-in o Shortcuts o Pickup o Meet me conference
• Turret Proxy to Open Line Dealing Server (TPO): The TPO server serves as a proxy
phone between a remote place extension and the local IP Trade end-users (turrets). When the local IP Trade end-users dial to the TPO proxy number of a remote place, the users are able to speak publicly or privately to that remote place. In these Application Notes, the TPO server registers one SIP endpoint as the proxy phone with Avaya SIP Enablement Services. One Avaya IP extension was used for remote place, in this case, Paris.
2. Test Configuration Figure 1 illustrates the configuration used to verify these Application Notes. The Main Location consists of an Avaya SIP Enablement Services server, Avaya Communication Manager (on an Avaya S8300 Media Server), an Avaya G350 Media Gateway, Avaya 4600 series IP Telephones, Avaya 6400 series Digital Telephones, IP Trade Profile IK turrets, an IP Trade TSS server, and an IP Trade TPO server. Note: Please consult the appropriate documentation listed in the References section of this document for more information on how to set up these components. The focus of these Application Notes is to configure the IP Trade Profile-IK with Avaya SIP Enablement Services and Avaya Communication Manager at the Main Location. IP Trade Call Dispatcher/Huntgroup was tested. Refer to [6] for detailed information on how to configure an IP Trade Dispatcher/Huntgroup.
4. Configure Avaya Communication Manager This section details the administration on Avaya Communication Manager to integrate with Avaya SIP Enablement Services and to enable the IP Trade Turrets to register as SIP endpoints. The commands listed in the following sections were issued at the Avaya System Access Terminal (SAT) to configure the Avaya S8300 Media Server. A SIP network interface must be configured between Avaya Communication Manager and Avaya SIP Enablement Services. This interface is a trunk group that handles all SIP signaling between Avaya SIP Enablement Services (which interfaces with the IP Trade Profile-IK as a SIP proxy) and Avaya Communication Manager. The steps described below enable the features and create the administrative objects necessary to support this interface.
Step Description
1. Enter display system-parameters customer-options to verify that sufficient SIP trunk capacity exists. On Page 2, under the IP PORT CAPACITIES section, verify that the Maximum Administered SIP Trunks is enough to support the expected traffic to and from each IP Trade Profile-IK. Any call involving a SIP endpoint will use a SIP trunk per SIP endpoint. If the capacity indicated is deemed insufficient, an authorized Avaya support technician will need to install an appropriately enabled license file. display system-parameters customer-options Page 2 of 10 OPTIONAL FEATURES IP PORT CAPACITIES USED Maximum Administered H.323 Trunks: 24 6 Maximum Concurrently Registered IP Stations: 24 1 Maximum Administered Remote Office Trunks: 0 0 Maximum Concurrently Registered Remote Office Stations: 0 0 Maximum Concurrently Registered IP eCons: 0 0 Max Concur Registered Unauthenticated H.323 Stations: 0 0 Maximum Video Capable H.323 Stations: 0 0 Maximum Video Capable IP Softphones: 0 0 Maximum Administered SIP Trunks: 60 24 Maximum Number of DS1 Boards with Echo Cancellation: 0 0 Maximum TN2501 VAL Boards: 1 0 Maximum G250/G350/G700 VAL Sources: 1 1 Maximum TN2602 Boards with 80 VoIP Channels: 0 0 Maximum TN2602 Boards with 320 VoIP Channels: 0 0 Maximum Number of Expanded Meet-me Conference Ports: 0 0 (NOTE: You must logoff & login to effect the permission changes.)
2. Enter change node-names ip to add the name (in this example, ses) and IP address (in this example, 192.168.1.3) for Avaya SIP Enablement Services. Note also the administered Avaya S8300 Media Server interface (in this example, procr), Audix (in this example, msgserver), and the Avaya G350 Media Gateway (192.168.1.254). The procr will be used in Section 5, Step 5 to create the Media Server Interface in Avaya SIP Enablement Services. change node-names ip Page 1 of 1 IP NODE NAMES Name IP Address Name IP Address default 0 .0 .0 .0 . . . procr 192.168.1 .1 . . . msgserver 192.168.1 .1 . . . ses 192.168.1 .3 . . . . . . . . . . . . . . .
3. Enter change ip-codec-set n (where n is the number of the codec set specified in Step 4 below) to specify the audio codecs to be used for calls routed to and from the IP Trade turrets via Avaya SIP Enablement Services. The order of the codecs listed will determine the negotiating preference for each call established. A prime consideration is whether voice quality is more important than bandwidth utilization (e.g., the higher bandwidth used by the G.711 codecs corresponds to better quality). The codecs must be among those supported by the IP Trade Profile-IK. change ip-codec-set 1 Page 1 of 2 IP Codec Set Codec Set: 1 Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.711MU n 2 20 2: G.711A n 2 20 3: G.729AB n 2 20 4: 5: 6: 7:
4. Enter change ip-network-region n, where n is the IP network region where the Avaya SIP Enablement Services server will reside, to define the connectivity settings for all VoIP resources and IP endpoints within that region. In this example, region 1, the default region for the Media Server running Avaya Communication Manager, was used. The following fields should be considered:
• Authoritative Domain: Enter a value that matches the SIP Domain of the Avaya SIP Enablement Services server (in this example, stone.com).
• Intra-region IP-IP Direct Audio, Inter-region IP-IP Direct Audio: Keep the default value of yes for each of these fields to allow audio traffic to be sent directly between IP endpoints without using media resources in the Avaya G350 Media Gateway (a feature known as “shuffling”).
• Codec Set: Enter the IP codec set specified in Step 3. This determines the set of codecs to be used for calls within this IP network region. Since the Avaya S8300 Media Server and the Avaya SIP Enablement Services server are in the same region in this example, all calls to and from any IP phone (H.323 or SIP) within the enterprise will use this codec set. If the Avaya S8300 Media Server and the Avaya SIP Enablement Services server are placed in separate regions, then calls between trading turrets connected to the IP Trade and other IP phones (such as the Avaya 4600 Series IP telephones in this example) would be subject to the codec set specifications on Page 3 of the corresponding IP Network Region forms.
change ip-network-region 1 Page 1 of 19 IP NETWORK REGION Region: 1 Location: 1 Authoritative Domain: stone.com Name: MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 1 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? y UDP Port Max: 65531 DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS Audio PHB Value: 46 Use Default Server Parameters? y Video PHB Value: 26 802.1P/Q PARAMETERS Call Control 802.1p Priority: 6 Audio 802.1p Priority: 6 Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS H.323 IP ENDPOINTS RSVP Enabled? n H.323 Link Bounce Recovery? y Idle Traffic Interval (sec): 20 Keep-Alive Interval (sec): 5 Keep-Alive Count: 5
5. Enter add signaling-group n to create a new SIP signaling group (to be used by the SIP trunk group to be created in Step 6). In this example, signaling group 1 was created. The following fields should be considered:
• Group Type: Enter sip. • Near-end Node Name: Enter the node name supporting the Avaya S8300 Media
Server (in this example, procr). For Media Server platforms that use C-LAN boards, node name corresponding to C-LAN would be specified here.
• Far-end Node Name: Enter the node name for the Avaya SIP Enablement Services server (in this example, ses).
• Far-end Listen Port: Enter 5061 (the recommended TLS port value). • Far-end Network Region: Enter the IP Network Region from Step 4. This
determines which IP network region contains the Avaya SIP Enablement Services server.
• Far-end Domain: Enter the domain name of the Avaya SIP Enablement Services server (in this example, stone.com).
• DTMF over IP: Enter rtp-payload. This value enables Avaya Communication Manager to send DTMF tones using RFC 2833.
• Direct IP-IP Audio Connections: Enter y to enable shuffling between the near-end and far-end IP endpoints.
add signaling-group 1 Page 1 of 1 SIGNALING GROUP Group Number: 1 Group Type: sip Transport Method: tls Near-end Node Name: procr Far-end Node Name: ses Near-end Listen Port: 5061 Far-end Listen Port: 5061 Far-end Network Region: 1 Far-end Domain: stone.com Bypass If IP Threshold Exceeded? y DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y IP Audio Hairpinning? y Session Establishment Timer(min): 120
6. Enter add trunk-group n to create a new SIP trunk group. In this example, trunk group 1 was created. On Page 1, enter the following values:
• Group Type: Enter sip. • Group Name: Enter a descriptive name. • TAC: Enter a valid trunk access code. • Service Type: Enter tie. • Signaling Group: Enter the number of the signaling group created in Step 5. • Number of Members: Enter an appropriate number of SIP trunks, not exceeding
the maximum number of available SIP trunks as indicated in Step 1. add trunk-group 1 Page 1 of 21 TRUNK GROUP Group Number: 1 Group Type: sip CDR Reports: y Group Name: to ses COR: 1 TN: 1 TAC: 101 Direction: two-way Outgoing Display? n Dial Access? n Night Service: Queue Length: 0 Service Type: tie Auth Code? n Signaling Group: 1 Number of Members: 24
On Page 3, specify the following value:
• Numbering Format: Enter public. This determines the outgoing calling party number format.
• Retain the default values for the other fields. change trunk-group 1 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: none Maintenance Tests? y Numbering Format: public Prepend '+' to Calling Number? n Replace Unavailable Numbers? n
7. All SIP stations are configured as OPS stations on Avaya Communication Manager. This includes the IP Trade trading turrets and SIP endpoints used by the TPO.
Enter display system-parameters customer-options and examine Page 1 to confirm that the license file has allocated enough OPS extensions (Maximum Off-PBX Telephones – OPS) to support all enterprise sites. If not, an authorized Avaya support technician will need to install an appropriately enabled license file. display system-parameters customer-options Page 1 of 10 OPTIONAL FEATURES G3 Version: V13 Location: 1 RFA System ID (SID): 1 Platform: 13 RFA Module ID (MID): 1 USED Platform Maximum Ports: 900 131 Maximum Stations: 450 41 Maximum XMOBILE Stations: 0 0 Maximum Off-PBX Telephones - EC500: 50 0 Maximum Off-PBX Telephones - OPS: 60 24 Maximum Off-PBX Telephones - SCCAN: 0 0 (NOTE: You must logoff & login to effect the permission changes.)
8. Enter add station x (where x is an available valid extension in the dial plan) to create a station extension for a SIP endpoint. Enter the following values on Page 1:
• Type: Set to 6408D+ (the default). • Port: Enter X. This indicates that the station is Administered Without Hardware
(AWOH), i.e., not assigned to a specific port on Avaya Communication Manager. • Name: Enter a descriptive name.
add station 31001 Page 1 of 4 STATION Extension: 31001 Lock Messages? n BCC: 0 Type: 6408D+ Security Code: TN: 1 Port: X Coverage Path 1: COR: 1 Name: IP-Trade 1 Coverage Path 2: COS: 1 Hunt-to Station: STATION OPTIONS Loss Group: 2 Personalized Ringing Pattern: 2 Data Module? n Message Lamp Ext: 30001 Speakerphone: 2-way Mute Button Enabled? y Display Language: english Media Complex Ext: IP SoftPhone? N
9. On Page 2 of this Station form, enter the following value:
• Direct IP-IP Audio Connections: Enter y to enable shuffling calls involving this station.
• Restrict Last Appearance: Enter n to allow the last call appearance to be used for either an incoming or outgoing call.
add station 31001 Page 2 of 4 STATION FEATURE OPTIONS LWC Reception: spe Auto Select Any Idle Appearance? n LWC Activation? y Coverage Msg Retrieval? y LWC Log External Calls? n Auto Answer: none CDR Privacy? n Data Restriction? n Redirect Notification? y Idle Appearance Preference? n Per Button Ring Control? n Bridged Idle Line Preference? n Bridged Call Alerting? y Restrict Last Appearance? n Active Station Ringing: single Conf/Trans on Primary Appearance? n H.320 Conversion? n Per Station CPN - Send Calling Number? Service Link Mode: as-needed Multimedia Mode: basic MWI Served User Type: Display Client Redirection? n AUDIX Name: Select Last Used Appearance? n Coverage After Forwarding? s Direct IP-IP Audio Connections? y Emergency Location Ext: 31001 IP Audio Hairpinning? y
10. On Page 3 of this Station form, enter the following BUTTON ASSIGNMENTS:
• Add at least two call appearances (call-appr) to enable the station to originate a second call, carry out a call transfer, etc. Note in this sample configuration, six call appearances were added for each SIP station used by the IP Trade turrets.
add station 31001 Page 3 of 4 STATION SITE DATA Room: Headset? n Jack: Speaker? n Cable: Mounting: d Floor: Cord Length: 0 Building: Set Color: ABBREVIATED DIALING List1: group 1 List2: List3: BUTTON ASSIGNMENTS 1: call-appr 5: call-appr 2: call-appr 6: call-appr 3: call-appr 7: 4: call-appr 8:
11. Enter add off-pbx-telephone station-mapping to map the new station extension to an OPS station. Enter the following values in the first available row:
• Station Extension: Enter the extension of the station created in Step 8. • Application: Enter OPS. • Phone Number: Enter the phone number of the associated Off-PBX Telephone. • Trunk Selection: Enter the number of the SIP trunk group created in Step 6 (in
this example, 1). • Configuration Set: Retain the default value.
add off-pbx-telephone station-mapping 31001 Page 1 of 2 STATIONS WITH OFF-PBX TELEPHONE INTEGRATION Station Application Dial Phone Number Trunk Configuration Extension Prefix Selection Set 31001 OPS - 31001 1 1 - - - -
12. On Page 2, set the Call Limit to the number of call appearances set on the station form in Step 10. Verify that the Mapping Mode is set to both.
add off-pbx-telephone station-mapping 31001 Page 1 of 2 STATIONS WITH OFF-PBX TELEPHONE INTEGRATION Station Call Mapping Calls Bridged Extension Limit Mode Allowed Calls 31001 6 both all both
13. Repeat Steps 8 - 12 as necessary to add additional phones.
14. Enter change off-pbx-telephone feature-name-extensions to administer the FNEs that will be used by the IP Trade turrets and the Avaya SIP endpoints to activate certain Avaya Communication Manager features. The FNEs shown in bold were the subset that were tested in the reference configuration. change off-pbx-telephone feature-name-extensions Page 1 of 1 EXTENSIONS TO CALL WHICH ACTIVATE FEATURES BY NAME Active Appearance Select: Idle : Appearance Select Automatic Call Back: Last Number Dialed: 3228 Automatic Call-Back Cancel: Malicious Call Trace: Call Forward All: 3213 Malicious Call Trace Cancel: Call Forward Busy/No Answer: 3214 Off-Pbx Call Enable: Call Forward Cancel: 3215 Off-Pbx Call Disable: Call Park: 3216 Priority Call: Call Park Answer Back: 3217 Send All Calls: 3234 Call Pick-Up: 3218 Send All Calls Cancel: 3235 Conference on Answer: 3219 Transfer On Hang-Up: 3236 Calling Number Block: Transfer to Voice Mail: 3237 Calling Number Unblock: Whisper Page Activation: Directed Call Pick-Up: 3222 Drop Last Added Party: 3223 Exclusion (Toggle On/Off): Extended Group Call Pickup: Held Appearance Select:
5. Configure Avaya SIP Enablement Services This section addresses the administrative steps to be performed on Avaya SIP Enablement Services. The installation of the Avaya SIP Enablement Services software and license file, as well as the initial configuration of the server, is beyond the scope of this document. Please see [3] for the details of these procedures.
Step Description
1. To administer Avaya SIP Enablement Services, navigate to http://<ip-addr>/admin (where <ip-addr> is the IP address of the Avaya SIP Enablement Services server) from a Web browser. After logging in with an appropriate login and password, the main page appears.
2. Click on the Launch Administration Web Interface link. The Administration Home Page appears. NOTE: After making each of the changes described in this section, use the Update link, found at the bottom of the blue navigation pane, to commit the changes to the Avaya SIP Enablement Services database.
3. From the left pane, select Server Configuration → System Properties. On the Edit System Properties page, enter the following values: SIP Domain: This must match the Authoritative Domain field configured on
the IP Network Region form in Avaya Communication Manager shown in Section 4, Step 4. In this example, stone.com is used.
License Host: Enter the host name, the fully qualified domain name, or the IP address of the WebLM server where the license file for this Avaya SIP Enablement Services server is installed. In this example, the WebLM server is co-resident with the Avaya SIP Enablement Services server (IP address 192.168.1.3).
The completed form appears as follows. Click Update to submit the form.
4. From the blue navigation pane, select Hosts, then select Edit on the next page. The Edit Host page appears. Enter the following values:
• Host IP Address: Enter the Logical IP or Logical Name (from the Edit System Properties form in Step 3).
• DB Password: This is the password that was entered during execution of the system installation script. (See [3] for details.)
• Host Type: This field indicates whether this Avaya SIP Enablement Services server functions as a “home” or “edge” server. Since, in this example, only a single Avaya SIP Enablement Services server is used, enter home/edge.
• Keep all other default values. The completed form appears as follows. Click Update to submit the form.
5. From the blue navigation pane, select Media Server → Add. The Add Media Server page appears. Enter the following values: Media Server Interface Name: Enter a descriptive name (in this example,
S8300). Host: Select the name or IP address of the Avaya SIP Enablement Services
server from the drop-down menu (in this example, 192.168.1.3). SIP Trunk Link Type: Select TLS (Transport Link Security). SIP Trunk IP Address: Enter the IP address of the S8300 Media Server from
Section 4, Step 2. Keep all other default values.
The completed form appears as follows. Click Add to submit the form.
6. SIP users (identified by the corresponding telephone extensions) must be added. From the blue navigation pane, select User → Add. Enter the following values: Primary Handle: This specifies a user in Avaya SIP Enablement Services (in
this example, 31001). While not required, it is recommended that the Primary handle be the same as the User ID.
User ID: Set the same value as Primary Handle. Password: This must match the password associated with the corresponding
User ID configured on the IP Trade for a turret in Section 6.1, Step 2 and 4. Confirm Password: Re-enter the above Password. Host: Select from the drop-down menu the host name or IP address of the
Avaya SIP Enablement Services server (in this example, 192.168.1.3). First Name, Last Name: Enter descriptive values. Add Media Server Extension: Check this box to add an OPS extension for the
user (see Step 8). The completed form appears as follows. Click Add to submit the form.
8. The Add Media Server Extension page will appear as follows. Enter the following values: Extension: Enter the OPS extension corresponding to the station configured in
Avaya Communication Manager. See Section 4, Step 11 - 12. Media Server: Select the Media Server added in Step 5 (in this example,
S8300). The completed form appears as follows. Click Add to submit the form.
9. Repeat Steps 6-8 to add additional users to support the remaining IP Trade/SIP telephones.
11. To view the configured extensions, select Media Server Extensions → List from the blue navigation pane.
12. A Media Server Address Map must be administered to enable FAC, FNE, and calls to non-SIP endpoints to be routed to the appropriate Avaya Communication Manager. SIP call requests are made via an INVITE message, which includes as its destination a Uniform Resource Identifier (URI). Generally, the URI is of the form
sip:<user>@<domain> where <domain> is either a domain name or an IP address. For the reference configuration, a SIP URI in an INVITE message specifying the user created in Step 6 would look like the following:
sip:[email protected] Use the following steps to configure a Media Server Address Map for extensions beginning with a 4: From the blue navigation pane, select Media Servers → List. The List Media
Step Description Select the Map link associated with the appropriate Media Server. The List
Media Server Address Map page appears. Select the Add Map In New Group link. The Add Media Server Address
Map page appears. Enter the following values: o Name: Enter a descriptive value (in this example, nonesip-4xxxx). o Pattern: Enter the regular expression to be used for pattern matching.
In this example, pattern specification for station extensions is given: ^sip:4[0-9]{4}. This expression will match any URI that begins with the text string “sip:4” (the ^ matches the beginning of the line) and ending with any other combination of 4 digits. (See [4] for more details on address map syntax.)
o Replace URI: Keep the default (√). The completed form appears as follows. Click Add to submit the form.
13. After submitting the previous form, the List Media Server Address Map page appears as follows, showing an entry for the newly created Address Map.
In addition, a Media Server Contact is created automatically after the first Media Server Address Map is added. The contact specifies the IP address of the Avaya S8300 Media Server (in this example, 192.168.1.1) and the transport protocol used to send SIP signaling messages. The user in the original request URI is substituted for $(user).
6. Configure IP Trade System This section addresses the administrative steps to be performed on the IP Trade system. The installation of the IP Trade system software, as well as the initial configuration of the turrets and servers, is beyond the scope of this document. Please see [5] for the details of these procedures.
6.1. Configure IP Trade Turret Support System (TSS) This section describes the procedure for configuring the IP Trade TSS server. This procedure assumes that the TSS has already been configured with an anonymous profile, the IP addresses for TSS, Avaya S8300 Media Server, Avaya G350 Media Gateway, and Avaya SIP Enablement Services.
Step Description
1. To administer IP Trade TSS, navigate to http://<ip-addr>/iptradenet.console (where <ip-addr> is the IP address of the TSS from a Web browser. After logging in with an appropriate login and password, the main page appears.
2. To add a new turret device, click Telephony Accounts and select Turret Devices. Click Add new. The Turret Device Edition screen appears. Enter the following values:
• Turret Identifier: Enter the MAC address of the turret. • SIP Display Name: Enter a descriptive name (in this example, anonymous1). • SIP Extension: Enter the extension number administered in Section 5, Step 6 (in
this example, 31001). • SIP Password: Enter the password administered in Section 5, Step 6. • SIP Digest: Enter the extension number entered in the SIP Extension field. • SIP Domain: This must match the Authoritative Domain field configured on
the IP Network Region form in Avaya Communication Manager shown in Section 4, Step 4. In this example, stone.com is used.
3. To add a user profile, click Telephony Accounts and select User Profiles. Click Add new. The User General Edition screen appears. Enter the following values:
• First Name: Enter the first name of the user (in this example, User1). • Last Name: Enter a descriptive name (in this example, User1). • Profile Name: Enter the name of the profile (in this example, User1). • Status: Select Active. • Role: Select Users.
5. The User Authentication Edition screen appears. Enter the following values:
• User Identity: Enter the user identity (in this example, U1). • Login Password: Enter the password. • Confirm Login Password: Enter the password again • SIP Display Name: Enter the display name of the user (in this example, User1). • SIP Extension: Enter the extension number assigned to this user, (in this
example, 31001). The Station is administered in Section 5, Step 6. • SIP Password: Enter the password of the extension entered in Section 5, Step 6. • SIP Digest: Enter the extension number. • SIP Domain: This must match the Authoritative Domain field configured on
the IP Network Region form in Avaya Communication Manager shown in Section 4, Step 4. In this example, stone.com is used.
• Use defaults for all other values
6. Repeat Steps 3 - 5 as necessary to add additional user profiles.
7. The global shortcut is used by the turret for speed dial. To add a global shortcut, click Shared Database, and then Global Shortcuts. The Global Shortcut Edition screen appears. Enter the following values:
• Name: Enter a descriptive name (in this example, Avaya1). • Device: This option allows user to select the device on which the call will be
done. In this example, select No Device for Avaya phones. Note: If Handset 1 is selected for the shortcut Avaya1, when this shortcut is used to place a call, the call will be placed on Handset 1.
• Type: For turrets or Avaya phones, select Extension. • Extension: Enter the extension number assigned to the name entered above.
8. Repeat Step 7 to add IP Trade turrets. In this example, either handset 1 or handset 2 can be used by the IP Trade turret. In the Device drop down menu, select either Hand Set 1 or Hand Set 2.
9. Repeat Step 7 to add an open line for remote place. In this example, Loudspeaker is used for the Open Line Dealing (OLD). In the Device drop down menu, select Loudspeaker. In the Type field, select OLD.
6.2. Configure the IP Trade Turret This section describes the procedure for configuring the IP Trade turret. When the turret is in the Maintenance mode, it can be accessed for administering the turret system options. The commands listed in the following sections were issued at the IP Trade turret GUI interface.
Step Description
1. To administer IP Trade turret, log in to the turret with appropriate credentials. Click Turret settings.
4. To enter the URI of the TSS, click Use following TSS location. In the Primary server URI field enter the URI of the TSS (in this example, http://192.168.1.211/Iptradenet.TSS).
5. Click the Turret Options tab, and then click Use the turret MAC address. To specify the audio codecs to be used, select the IP codecs G711u, G.711a, and G729 from the Supported CODECs column and move to the Announced CODECs column.
6.3. Configure Turret Proxy to Open Line Dealing (TPO) The TPO server reads the TPO configuration file to run the TPO service. This configuration file is maintained by the IP Trade system. The fields modified for this sample configuration are shown in Step 1.
Step Description
1. place.names = Paris,Tokyo
#The extension numbers for the remote place, in this example, 40000. place.Paris.extension = permanent, 40000 place.Paris.conferencemode = all
#The SIP extension number used as the proxy number, in this example, 31007. place.Paris.localextensions = 31007 place.Paris.31007.localdigest = 31007 place.Paris.31007.localpwd = ***** # The IP addresses of the TPO server application.sip.connection.ipaddress = 192.168.1.210 # Avaya SIP server Information application.sip.server.name = 192.168.1.3 application.sip.server.port = 5060 application.sip.server.expire = 30 application.sip.server.registeringbeforeexpire = 7 application.sip.connection.mode = UDP # tcp/tls persistency, 3 possible values: none, transaction,transactionuser application.sip.connection.persistency = none application.sip.connection.port = 5060 application.sip.server.mediagateway = 192.168.1.254 # if the proxy.hostname config is set , the ipaddress is not necessapplication.sip.proxy.address = 192.168.1.3
7. Interoperability compliance Testing This section describes the interoperability compliance testing used to verify the interoperability between IP Trade and Avaya SIP Enablement Services with Avaya Communication Manager. This section covers the general test approach and the test results.
7.1. General Test Approach The following features and functionality were covered during the IP Trade interoperability compliance test. All calls involved various Avaya telephones and the IP Trade turrets.
• Outgoing calls from IP Trade Profile IK turrets to Avaya telephones • Incoming calls from Avaya telephones to IP Trade Profile IK turrets • G.711MU, G.711A, and G.729AB codecs for voice calls • DTMF tone transmission with successful Voice Mail access • Telephone features such as hold, transfer, conference, and voice mail • IP Trade Profile IK turret can join meet-me conference • IP Trade Profile IK turrets call handling panel uses Avaya Communication Manager
feature name extensions (FNE) • Direct IP-to-IP media (also know as “shuffling”) with SIP telephones
7.2. Test Results Interoperability testing of the sample configuration was completed with successful results. The following observations were made during testing:
1. The IP Trade turret can not work with the Feature Name Extensions (FNE), if the IP Trade Intercom is enabled. The work around is to disable the IP Trade Intercom.
2. If the Meet-me conference is configured with password (PIN), the intercom feature will not work. The work around is to remove the announcement and collect password steps from the vector.
3. The IP Trade Barge-in feature does not provide warning beep tone. 4. The IP Trade turret can not work with the Feature Name Extensions (FNE), if the IP
Trade Intercom is enabled. The work around is to disable the IP Trade Intercom.
IP Trade and Avaya are planning to work on a solution to make the missing features available. Until then, the Solution can be released only on a project basis with individual approval from Avaya´s Industry Solutions Management. Please contact Markus Blonski for discussion and information (mail to: [email protected]).
8. Verification Steps The following steps can be used to verify that the configuration steps documented in these Application Notes have been done correctly.
• Verify IP Trade turrets register with Avaya SIP Enablement Services, select User → Registered Users on the Avaya SES administrative web interface.
• Verify that calls can be placed between two IP Trade turrets for at least 2 minutes. • Verify IP Trade turrets can place calls to Avaya telephones and that the calls remain
connected for 2 minutes. • Verify IP Trade turrets can receive calls from Avaya telephones and that the calls remain
connected for 2 minutes. • Verify that all above calls are properly disconnected when either end disconnects first. • Using the list trace station command, verify that voice calls are using the expected codec. • Verify that a call can be placed between an IP Trade turret and an Avaya 4600 Series
H.323 IP telephone.
9. Support Technical support for the IP Trade Profile-IK can be obtained from IP Trade. See the Support link at www.IPTrade-networks.com for contact information.
10. Conclusion The IP Trade Turrets can successfully register to Avaya SIP Enablement Services and utilize telephony features of Avaya Communication Manager.
11. Additional References The following Avaya product documentation can be found at http://support.avaya.com. [1] Feature Description and Implementation For Avaya Communication Manager, Issue 4.0, February 2006, Document Number 555-245-205. [2] Administrator Guide for Avaya Communication Manager, Issue 2.1, May 2006, Document Number 03-300509. [3] Installing and Administering SIP Enablement Services, March 2007, Issue 2.1, Document Number 03-600768. [4] SIP Support in Release 3.1 of Avaya Communication Manager Running on the Avaya S8300, S8300, S8300B, S8700, and S8710 Media Server, February 2006, Issue 6, Document Number 555-245-206. The following IP Trade product documentation is available from IP Trade. Visit http://www.IP Trade-networks.com for company and product information. [5] IP Phone for Trading Room Applicative Layer Turret Installation Release 2.0. [6] IP Phone for Trading Room Applicative Layer TSS Installation Release 1.0.
