124i Enhanced 4 VoIP PCB. Installation Manual

124i Enhanced 4 VoIP PCB Installation Manual 20100

This manual has been developed by NEC America. It is intended for the use of its customers and service personnel, and should be read in its entirety before attempting to install or program the system. Any comments or suggestions for improving this manual would be appreciated. Forward your remarks to: NEC America, Inc., Corporate Networks Group 4 Forest Parkway, Shelton, CT 06484 cng.nec.com Nothing contained in this manual shall be deemed to be, and this manual does not constitute, a warranty of, or representation with respect to, any of the Equipment covered. This manual is subject to change without notice and NEC America has no obligation to provide any updates or corrections to this manual. Further, NEC America also reserves the right, without prior notice, to make changes in equipment design or components as it deems appropriate. No representation is made that this manual is complete or accurate in all respects and NEC America shall not be liable for any errors or omissions. In no event shall NEC America be liable for any incidental or consequential damages in connection with the use of this manual. This document contains proprietary information that is protected by copyright. All rights are reserved. No part of this document may be photocopied or reproduced without prior written consent of NEC America. 2003 by NEC America, Inc. All Rights Reserved Printed in U.S.A.

Table of Contents 4VoIP PCB Overview.......................................... 1 Hardware Considerations...................................... 2 Programming Considerations................................... 2 Outgoing Calls If Using a Gatekeeper........................... 2 Outgoing Calls If Not Using a Gatekeeper........................ 2 Configuration for IP Address Determination............................. 3 How to Determine the Destination IP Address with ARS (Required if Dialing More Than 4 Digits)...................................... 3 Dial Edit......................................................... 5 Calling Party Number.............................................. 5 F-Route Setting for IP Address.................................. 5 Incoming Calls............................................... 6 DID Error Routing Settings.......................................... 6 VoIP PCB Setup Examples...................................... 7 Conditions................................................... 8 Installation................................................... 8 Programming................................................. 9 Firmware Upgrade........................................... 16 92046INS01 Table of Contents- 1

Table of Contents Table of Contents - 2 92046INS01

4VoIP PCB Overview The 4 port VoIP PCB (P/N 92046) provide a means of connecting multiple i-series systems together over an IP WAN. With the addition of the VoIP PCB and using 124i Enhanced software version 2.01.00 or higher, it is now possible to place calls between i-series systems using VoIP and the Digital Networking features. Transmission of both voice and data over the same connection is now possible. This eliminates the need for point-to-point connection between sites and allows the use of a shared public network such as Frame Relay or ATM. Currently, the VoIP feature can provide: Desk-to-desk calls between i-series systems - Trunk-to-trunk connections between sites - Only DID call handling is supported Ability to transfer intercom or trunk calls to an extension on another i-series system Ability to transfer intercom calls from a remote site to a local trunk on the system Direct access or dial access to trunks at another i-series site i-series Networking features (such as common voice mail with voice message indication) Release Link Tie Trunk Transfer Trunk release after transferring a call to another i-series site An extension user at one site can easily dial an extension at a remote i-series telephone system. Calls are transported between systems using the H.323 protocol with ISDN-like messages that contain information about the source of the call, the type of call and the destination of the call. This call information is sent within a User-User Information element which is part of an H.223/H.225 packet. VoIP call routing and control may be done with an external H.323 Gatekeeper. Each client is required to register with an H.323 Gatekeeper and provide an IP address and telephone number to be used as an alias. From this information, the Gatekeeper constructs routing tables and provides clients with directory services. When a Gatekeeper is installed, all routing and call initialization must be handled by the Gatekeeper. The 124i Enhanced system will not be allowed to specify the destination IP address for a call if a Gatekeeper is installed. When a Gatekeeper is not available, the 124i Enhanced must examine the dialed numbers and determine a routing method which may include an IP address and an outgoing trunk group. The 124i Enhanced must be configured to use either F-Route or ARS to translate a dialed number into an IP address and select a route for a VoIP call. A maximum of 5 VoIP PCBs may be installed in a system, but with the following limitations. The first cabinet can only have one VoIP PCB installed and must be in slots 4-8. The second and third cabinets can each have only two VoIP PCBs installed and must also be installed in the last four slots in the cabinet (slots 12-16 in the second cabinet, slots 20-24 in the third cabinet). All trunks ports on the same VoIP PCB must be assigned to the same trunk group since incoming VoIP calls will select any idle port on the PCB. The VoIP PCB provides the following: Run/Block switch for enabling/disabling the PCB Load switch used for updating the PCB firmware PCMCIA card slot for updating PCB firmware DIN connector for diagnostics (no DIMU is required, simply plug a terminal into the connector) 8-pin LAN connector (CN10) Status LEDs for each VoIP port Run LED indicates if PCB is functioning Run/Block Switch Load Switch Reset Switch Run LED Status LEDs VoIP PCB CN10 Connector DIN Connector Page 1

Hardware Considerations Each VoIP PCB must be associated with a VoIP PCB at another i-series site. There must be a dedicated one-to-one relationship between the VoIP PCBs across the network. For example, if a remote site has 2 VoIP PCBs, the main office must also have 2 PCBs for sending and receiving VoIP calls to that site. Outgoing calls on a VoIP PCB must be routed only to the IP address of the associated VoIP PCB at the remote site. This is done to emulate the point-to-point connection provided by a T1 or PRI trunk. With this configuration, if all ports on the local PCB are busy, then all ports on the associated remote VoIP PCB will also be busy. This is required in a system with multiple VoIP PCBs to force overflow of outgoing calls to an alternate route (another VoIP PCB) if all ports at the remote site are busy. Programming Considerations Each VoIP PCB must be assigned to its own trunk group in Program 0905 - Trunk Groups. All ports on a VoIP PCB must be assigned to the same trunk group. When multiple VoIP PCBs are installed at a site, ARS or F-Route must be set up with alternate routes with each route sending calls to a different outgoing trunk group (VoIP PCB). Outgoing calls to a particular VoIP PCB must use only the IP address of the associated VoIP PCB at the remote site. It is recommended that the hunting order (the order of the alternate route selection) be reversed at each site to reduce the chance of call collisions. Outgoing Calls If Using a Gatekeeper If a Gatekeeper is used, the 124i Enhanced must be programmed to use the Gatekeeper. All user-dialed digits for a VoIP call are sent to the Gatekeeper which will set up and manage the call. Some Gatekeepers accept overlap dial information - some do not. The 124i Enhanced system can enable either Overlap or En Bloc Sending for each VoIP PCB (Program 0319 Item 1). This results in differences in the operation of outgoing calls. Overlap Sending Each time a user dials a digit, the dial information is sent to the Gatekeeper. This operation is just like using an analog trunk. En Bloc Sending Dial information must be stored and then sent all at once. After dialing the digits for an outgoing call, the user must wait for the inter-digit timer to expire before the digits are sent. The user may, however, press the # key to send the dialed digits before the timer expires. Outgoing Calls If Not Using a Gatekeeper A user is not allowed to seize the VoIP trunk manually because dialed digits cannot be sent directly to the VoIP trunk. An IP address must be provided to route the call. This means that a user cannot press a trunk key or dial a trunk access code to seize a VoIP trunk. To determine the destination IP address from the dialed digits, the 124i Enhanced system uses F-Route or ARS. These programs have been modified to support VoIP. The following description shows how to determine an IP address. How to Determine the Destination IP Address Using ARS or F-Route This option must be used for reaching destinations such as a key telephone at a remote system without a Gatekeeper. The target destination will be specified by a local trunk group number, an IP address and the called party number. Page 2

Configuration for IP Address Determination In Program 0905, program all the VoIP trunks into the same group. This trunk group can then be selected in Program 0928 as a route for a VoIP call. Program 0905 VoIP trunk port 1 2 3 4 5 6 7 8 VoIP trunk group 26 ARS and F-Route can select the trunk group from the dialed digits for outgoing calls. The trunk group is part of the routing instructions in Program 2101 (ARS Selection Table) or Program 2904 (F-Route Routing Table). If the trunk group number is between 101 and 126, it indicates that an attempt should be make to route the call over VoIP. The last 2 digits of this trunk group number are used to select an entry (table) in Program 0928. The system will choose one trunk from the VoIP trunk group and use the assigned IP address. The dialed digits are edited by the appropriate dial treatment and stored in the called party number area in the Setup message. Program 0928 is a table that assigns a VoIP trunk group and an IP address for a VoIP call. This address is the IP address for the VoIP PCB at the remote location. Program 0928 Table VoIP Trunk Group IP Address 1 26 192.168.0.1 2 26 192.168.0.2 3 26 192.168.0.3 As a result of the programming shown in the table above, when trunk group 101 is selected by F-Route or ARS, the system will choose an outgoing trunk port from trunk group 26 and the IP destination address used will be 192.168.0.1. How to Determine the Destination IP Address with ARS (Required if Dialing More Than 4 Digits) If you want to dial the phone number to reach an IP telephone when a Gatekeeper is not available, the ARS feature should be used. ARS uses the 3-digit or 6-digit table to decide the trunk group to be used. An option is now available to convert the dialed digits to an IP address. If the ARS selection number is found by searching the 3 or 6-digit tables and the trunk group number in the Selection Routing Table (Program 2101) is between 101 to 126, it indicates that the system should attempt to route the call using VoIP. The last 2 digits of this trunk group number are used as the entry in Program 0928 to get the VoIP trunk group number and IP address. If the VoIP trunk group number is NOT set to zero (0) in Program 0928, the system starts searching for the dialed digits in a section of the Abbreviated Dial Table within the range defined in Program 0605. If a match is found, the associated entry in Program 0604 is a trunk group number that points to a table in Program 0928, which is used to get the VoIP trunk group and IP address. If no match is found, the system uses the trunk group number in the Program 0928 table that was chosen by the ARS Routing Table. This may be a trunk group number for a normal trunk group (without an IP address) or a VoIP trunk group (with an IP address). Page 3

VoIP Using ARS and No Gatekeeper for Outgoing Calls Start Placing an outgoing call, a user dials 9 (trunk access code). ARS starts. The user dials a telephone number. The inter-digit timer expires or the user presses #. The system searches the ARS 6-Digit then the 3-Digit Table to find a selection number. The trunk group number from the Selection Route Table, Program 2101, is used for the call. The call is routed to the indicated trunk group. This is not a VoIP call. No Is this a VoIP trunk group number (101 to 126)? Yes Use the last 2 digits of the trunk group number as the entry point to Program 0928. Stop Yes Is the trunk group field in Program 0928 blank or set to '0'? No The system searches for the dialed digits in the Abbreviated Dial Table IP Area (Program 0603) within the search range (Program 0605). Is a match found? No Yes Does the table have a trunk group setting in Program 0604? The trunk group and IP address in Program 0928 are used to route the call. No Yes The VoIP call is routed on the indicated IP trunk using the IP address. Yes Is this a VoIP call (is there IP address data?) Use the trunk group as an entry point to Program 0928. No Stop The call is routed using the CO trunk group indicated. Page 4

VoIP Using F-Route and No Gatekeeper for Outgoing Calls Start To place an outgoing call, a user dials the F-Route code plus the telephone number. The system waits for the F-Route digits. Yes Does the outgoing trunk group have IP settings in Program 0928? No The system makes the call on a VoIP trunk with the edited number by F-Route. The system follows the F-Route programming or, if F-Route is not set, the call is not completed. Stop Stop Dial Edit ARS and F-Route have the ability to edit dialed digits. The edited dial digits are placed in the called party number area of the Setup message.this dialed digit editing may be required to match the numbering plan of the remote telephone equipment. Calling Party Number If you need to send the calling party number, use the following programs. Program 0406 Item 80 ISDN Calling Party Number Program1031 ISDN Calling Number for Extensions Program 0925 Calling Number for ISDN Trunks F-Route Setting for IP Address The following example explains how to set up F-Route. With this programming, you can dial 401 to access another node and the IP address would be 192.178.10.2. Program the VoIP PCB Operating Mode In 0319, assign the slot number in which the VoIP PCB is installed. Program the Hardware Settings for the VoIP. In 0153, set the following: Item 1: IP Address: System A = 192.178.10.1 System B = 192.178.10.2 Subnet Mask: 255.255.0.0 Default Gateway Address: 192.178.10.10 Item 5: Fast Connect Mode: Fast Connect (Mode 2) Program the VoIP Trunk Group. In Program 0905, assign VoIP trunks to the same trunk group 10. Page 5

Program the VoIP Trunk Group and IP address for Trunk Groups. In Program 0928, set trunk group 1 for routing to a VoIP trunk. Table (Entry Number) 1 VoIP Trunk Group Number: 10 IP Address: 192.178.10.2 Program F-Route. Program 0501 Dial 4 1 digit Type 9 (F-Route) Program 2902 Flexible Numbering Plan Lookup Table Dial 4 3 digits Entry No.1 Entry Number is defined by Program 2903. Program 2903 Flexible Route Selection Table Entry No.1 Route Table No. 1 Route Table Number is defined by Program 2904. Program 2904 Flexible Routing Tables VoIP Route Table No. 1 Trunk Group 101 VoIP Trunk Group 101 will point to Program 0928, table (entry) 1. As a result, if a user dials the VoIP number 401, trunk group 101 is referred by F-Route. Then, the system checks the Program 0928 setting and seizes a trunk from system trunk group 10. The call goes to address 192.178.10.2. Special programming will be required if a remote system has two or more VoIP PCBs. Incoming Calls When an incoming call is detected, the VoIP PCB will be searched for an idle port on the VoIP PCB. An incoming call can be connected to any of the ports on the PCB, so the trunk service type has to be the same for all the trunks on the same PCB. Basic system settings are essentially the same as ISDN trunks. The routing programing is also the same. However, as incoming calls may have a called party number, it is recommended that DID service type be used (Program 0901, Items 14-17 and 32-35 set to 3). Incoming calls may not have a called party number or the number may not exist in the DID table. When this occurs, the DID Error Routing will be used. Refer to the i-series Software Manual, P/N 92000SWG**, for information on programming DID trunks. DID Error Routing Settings In the past, if a called party number did not exist in the Setup message, ISDN DID trunks would reject the calls. With the VoIP H.323 incoming calls, DID error routing is used to handle the Setup message for calls without a called party number. For example, DID Table Area #1 is being used, routing to trunk ring group 10. Check the first DID table number in the DID table area (Program 1805). The first table is 0001. Program 1 for the DID intercepted call vacant number option (Program 1810). Program 1810 Table #0001 -> Set 1 for the item 1 Program the DID Intercept Ring Group (Program 1809). Program 1809 DID Table Area #1 -> Set TRG 10 Page 6

DID Incoming Routing with VoIP Start The system detects an incoming call. Does the Setup message contain a called number? No Yes Is there a Called Party Number in the DID table (Program 1805)? No Should the call follow DID Error Routing (Program 1810)? No Yes Yes Normal DID routing occurs. The system follows Program 1809 intercept ring group routing. Warning tone is heard. Stop VoIP PCB Setup Examples The following are examples of connection and setup. It is recommended, however, to consult with your MIS network administrator regarding IP address settings before using the VoIPU. Program 0319: Operating Mode VoIPU Setup 1 - NT mode Program 0153:1 IP Address 128.10.0.1 Subnet Mask 255.255.0.0 Router (Default Gateway) 128.10.0.2 Program 0153:2 Gatekeeper connection mode Gatekeeper Address RAS Port 1 - Manual Setting 128.10.0.3 1719 Router Setup Router IP Address 128.10.0.2 Subnet Mask 255.255.0.0 Global IP Address Automatically assigned Page 7

IP Network VoIPU ISDN/Analog Trunk Public Network LAN Router DSTU Gate Keeper Conditions Page 8 Programmable Function Keys: Do not program a trunk key with a VoIP trunk port. If Gatekeeper is not used, F-Route or ARS must be used. If Gatekeeper is used, the 124i Enhanced system cannot specify an IP address. DTMF Relay: If you want to use DTMF tones on VoIP trunks, the calling party s system has to support User Input (H.245), otherwise the tone will not be detected. Example: DID/DISA trunks or voice mail DTMF relay mode must enabled in Program 0153, Item 8 for the VoIP Unit. F-Route codes do not work with Last Number Redial. F-Route does not work with automatic redial. SMDR prints out the Called Party Number as dialed number digits. If the dial treatment deletes the F-Route code, that F-Route code will not appear on the SMDR report. Program 0901 - Basic Trunk Port Setup (Part A), Item 3: CODEC Gain Type does not work for VoIP trunks. See Program 0154 instead. Echo-Canceller (DSP): The echo canceller will eliminate an echo by subtracting the predicted value from receive-gain level. If highway gain has been added, DSP could receive unpredictable gain level. If this happens, the echo cannot be eliminated. If the gain was added by some system outside of the 124i Enhanced, an echo may occur. This is unavoidable. It is recommended that the i-series system with a VOIPU PCB not be placed behind a firewall. Firewalls restrict access to TCP/UDP ports, some of which are utilized by the VOIPU. To ensure adequate speech quality, it is recommended that QoS (Quality of Service) is implemented on any routers or switching hubs in the VoIP network. Refer to the router s manual for further information. If the i-series system with a VOIPU PCB is installed behind a router with NAT (Network Address Translation) enabled, it is necessary to use H.323 masquerading. (The router must have H.323 Masquerading. The H.323 setup message contains the local IP address in a Q.931 UUS Information element. This local IP address in the UUS IE must be modified from the router. The router has to replace the local IP with the global IP. This feature is not available on all routers. Refer to your router s manual for further information.) In case the registering at a Gatekeeper is not located in the local network, the router has to masquerade the RAS protocol as well. Note that the misconfigured router may cause one-way speech path connections. Installation Main Equipment Key Telephones A Switching Hub with Auto-Negotiation Feature or 100M Switching Hub is strongly recommended. The VoIP PCB is Radvision compatible. A Gatekeeper, if used, must be Radvision compatible. Each installed VoIP PCB must be dedicated to a VoIP PCB at another site. Refer to Hardware Considerations on page 2. 1. Check the following LAN settings before installing the VoIP PCB: Local IP Address Subnet Mask Default Gateway (If Router is used) VoIP Unit does not support DHCP. 2. Define the VoIP PCB settings in Program 0319. 3. Attach a grounded wrist strap to your wrist and a grounded metal object (such as CEU ground). 4. Remove the cover from the common equipment cabinet by unscrewing the two front panel retaining screws.

5. Set the run/block switch DOWN on the VoIP PCB. 6. As a precautionary step, using Program 0006, Item 2, delete/uninstall the slot in which the VoIP PCB is about to be installed. 7. Plug the VoIP PCB into slots 4-8 in the main system cabinet, slots 12-16 in the second cabinet, or slots 20-24 in the third cabinet. Only 1 VoIP PCB can be installed in the first cabinet, 2 VoIP PCBs each can be installed in the second and third cabinets. 8. Connect the cable from the switching hub to the CN10 connector on the VoIP PCB using a CAT 5 cable. 9. Set the run/block switch UP. Before proceeding to Step 10, wait 30 seconds and verify that the LED starts to flash. This indicates that the board is operating normally. 10. Replace the front cover and tighten the two front panel retaining screws. 11. Program the system with all the required VoIP options. Programming When programming VoIP trunks for DID or tie line operation, refer to the i-series Software Manual, P/N 92000SWG**, for all required DID or tie line programs. Program 0006 - Slot Control To delete VoIP trunk ports, use Program 0006 to delete the PCB slot information. The trunk ports are then also deleted. Program 0007 System Report Port Setup The VoIP PCB will display in the system reports as VoIPU-T. The ports assigned to the PCB will be displayed as well. Program 0008 - Alarm Report Port Setup If the VoIP PCB is unplugged from the system cabinet, you ll see alarm report similar to other PCBs. For example: 0102 ERR 04/05/02 19:15 Board Install VoIPU-T 05 00 0102 REC 04/05/02 19:15 Board Install VoIPU-T 05 00 Program 0009 - Loop Back Testing When using telephone programming, you can test the hardware. In Program 0009, the following test can be performed. This menu item... Lets you... With this data Results Menu 11 ADLSC Loopback Test Perform a loop back test from the CEU to the ADLSC. The PCB must be in NT mode. 11 (for menu item) + HOLD + trunk port number + HOLD. If it is operating correctly, you ll see: Good! If it is not, you ll see: No Good! Or Time Out! Program 0153 - VoIP Hardware Setup Note: Each VoIPU PCB must be programmed. The default settings for options 4-20 should work well at default and do not need to be changed. If the VoIPU PCB is working, changing this option will cause the PCB to reset. VOIPU Index Number ( 1 5 ). Item No. Item Input Data Default 1 IP Address of the VoIP PCB xxx.xxx.xxx.xxx 0.0.0.0 Subnet Mask Default Gateway Address Setup Message TCP Port (xxx = 0-255) 1024-65535 0.0.0.0 0.0.0.0 1720 The default should not be changed or the operation of the VoIP PCB will be affected. RTP/RTCP UDP Port 1024-65518 The default should not be changed or the operation of the VoIP PCB will be affected. 10020 Page 9

Item No. Item Input Data Default 2 Gatekeeper Connection Mode 1: No Gatekeeper 2: Manual setting * 3: Automatic setting * Gatekeeper Address Required if the connection mode is set to 2 (Manual). Multi Cast Address * If set to 2 or 3 and a Gatekeeper cannot be found, the VoIP PCB will not start. xxx.xxx.xxx.xxx xxx: 0-255 xxx.xxx.xxx.xxx xxx: 0-255 The default should not be changed or the operation of the VoIP PCB will be affected. 1 0.0.0.0 224.0.1.41 RAS Port 1024-65535 1719 Multi Cast Port The default should not be changed or the 1718 operation of the VoIP PCB will be affected. Gatekeeper ID Up to 40 alphanumeric characters GK_ID1 Alias Address 0123456789 3 First Payload Compression Type (Transmit Voice Compression) Define the first priority payload when the VOIPU negotiates with the destination. 1: G.711 2: Reserved 3: G.729 4: G723.1 3 4 These settings (except for Payload Mode) affect the transmit compression type as well as the receive. (see Item 4 for option settings) Payload Settings 1-4 Payload Mode 1 (G.711) 0: Disable 1 (Receive Voice Compression) 1: Enable Define the availability of each 2 - Reserved --- - payload mode. If a specified 3 (G.729) 0: Disable 1 payload is defined as Disable 1: Enable, the VoIP Call is not 4 (G.723.1) 0: Disable 1 available with this payload. 1: Enable However, G.711 may not be set as Disable as it must be available for the VoIP call. VIF Size (Voice Frames per Packet) Define the packet size (byte) for voice. If a small value is used, the voice delay may be reduced, but the network traffic may be high. If a large value is defined, the voice delay and echo may increase, but the network traffic may be lower. 1 (G.711) 1: Reserved 2 2: 20ms (160byte) 3: 30ms (240byte) 2 - Reserved --- - 3 (G.729) 1: Reserved 2 2: 20ms (20 byte) 3: 30ms (30 byte) 4: 40ms (40 byte) 5: 50ms (50 byte) 6: 60ms (60 byte) 7: 70ms (70 byte) 8: 80ms (80 byte) 4 (G.723.1) 1: 30ms (24 byte) 2: 60ms (48 byte) 1 Page 10

Item No. Item Input Data Default 4 (cont d) Jitter Buffer Mode Define the buffer mode for the jitter of the voice packet. If Adaptive is selected, the buffering size will be automatically adjusted according to the signal delay. Adaptive Immediately is recommended. Jitter Buffer Min Delay Define the size for the Jitter Buffer. The disconnected voice may be reduced if the delay size is large. The Min Delay must be greater than or equal to the VIF size. Jitter Buffer Normal Delay Define the size for the Jitter Buffer. The disconnected voice may be reduced if the delay size is large. The Normal Delay must be greater than or equal to twice the VIF size. Jitter Buffer Max Delay Define the size for the Jitter Buffer. The disconnected voice may be reduced if the delay size is large. The Max Delay must be greater than or equal to four times the VIF size. Jitter Buffer Idle Noise Level Define the voice level of the idel packet (idle noise level) for VAD (Voice Activity Detection) configuration or Jitter Overflow (packet loss). It is recommended to define the minimum value. 5 Fast Connection Mode Determine whether the addess should be sent in the setup message reducing setup time (fast connect mode). 1 (G.711) 0: Disable 2 1: adaptive during silence period only 2: adaptive immediately 2 - Reserved --- - 3 (G.729) 0: Disable 2 1: adaptive during silence period only 2: adaptive immediately 4 (G.723.1) 0: Disable 1: adaptive during silence period only 2: adaptive immediately 2 1 (G.711) 1-145 (1 step 1ms) 20 2 - Reserved --- - 3 (G.729) 1-500 (1 step 1ms) 20 4 (G.723.1) 1-500 (1 step 1ms) 30 1 (G.711) N1-145 (1 step 1ms) 40 [ N1=The value of G.711 Min delay ] 2 - Reserved --- - 3 (G.729) N2-500 (1 step 1ms) 40 [ N2=The value of G.729 Min delay ] 4 (G.723.1) N3-500 (1 step 1ms) [ N3=The value of G.723.1 Min delay ] 60 1 (G.711) M1-145 (1 step 1ms) 80 [ M1=The value of G.711 Nor delay ] 2 - Reserved --- - 3 (G.729) M2-500 (1 step 1ms) 80 [ M2=The value of G.729 Nor delay ] 4 (G.723.1) M3-500 (1 step 1ms) 90 [ M3=The value of G.723.1 Nor delay ] 1 (G.711) 0-2000 (1 step 1dBm) 0 2 - Reserved --- - 3 (G.729) 0-2000 (1 step 1dBm) 0 4 (G.723.1) 0-2000 (1 step 1dBm) 0 dbm Entries: 0 = -70.00 1 = -69.99 2 = -69.98 3 = -69.97 4 = -69.96 1997 = -50.03 1998 = -50.02 1999 = -50.01 2000 = -50.00 1: Normal mode 2 2: Fast Connect mode Page 11

Item No. Item Input Data Default 6 Echo Canceller Mode 1: Enable 2: Disable Echo Canceller Tail size (Delay Volume) 1: 8ms 2: 16ms Echo Canceller Nonlinear Processor 1: Enable 2: Disable Echo Canceller NLP Comfort Noise Level 0-30 Entries: 0 = -70 1 = -69 2 = -68 3 = -67 28 = -42 29 = -41 30 = -40 Echo Canceller NLP Comfort Noise 0: Adaptive Configuration 1: fixed Echo Canceller 4 Wire-Detect 0: Disable 1: Enable 7 Silence Suppression VAD Configuration Detects the period of silence and send the Nil Packet to reduce the network traffic. Silence Suppression VAD Power Threshold Define the threshold of the nil voice level to detect the period of silence. 0: Disable 1: Enable 0 or 20-50 VAD Power Threshold Entries: 0 = Adaptive threshold 20 = -20 21 = -21 22 = -22 48 = -48 49 = -49 50 = -50 8 DTMF Relay Mode 0: Disable (In-Band Signaling) 1: Enable (Out-of-Band Signaling) DTMF Relay Type Define the type of DTMF Relay. With In-Band, the DTMF signal is on the extension header of the RTP (UDP), and it is sent to the destination. With Out-Band, the DTMF signal is sent to the destination with H.245 User Input (TCP). DTMF Low Band Amplitude Define the low band level of DTMF signals from the VOIPU to the PCM highway when the VOIPU receives the packet including DTMF information. Note: If the VoIPU PCB is working, changing this option will cause the PCB to reset. 1: In-bandwidth (Rfc2833) 2: Out-bandwidth (H.245) 0-830 Low Band Amplitude Entries: 0 = -80.0 1 = -79.9 2 = -79.8 3 = -79.7 800 = 0 828 = 2.8 829 = 2.9 830 = 3.0 1 2 1 0 0 1 0 0 1 2 670 Page 12

Item No. Item Input Data Default 8 (cont d) DTMF High Band Amplitude Define the high band level of DTMF signals from the VOIPU to the PCM highway when the VOIPU receives the packet including DTMF information. Note: If the VoIPU PCB is working, changing this option will cause the PCB to reset. 0-830 690 High Band Amplitude Entries: 0 = -80.0 1 = -79.9 2 = -79.8 3 = -79.7 800 = 0 828 = 2.8 829 = 2.9 830 = 3.0 1-3 1 9 Timer & Counter Setup Table Number Assign the table (defined in 0155) 10 FTP Server Address xxx.xxx.xxx.xxx xxx: 0-255 0.0.0.0 FTP Password Up to 8 alphanumeric characters username 11 - Reserved Item - ---- 0 12 - Reserved Item - ---- 0 13 - Reserved Item - ---- 0 14 - Reserved Item - ---- 0 15 - Reserved Item - ---- 0 16 - Reserved Item - ---- 0 17 - Reserved Item - ---- 0 18 - Reserved Item - ---- 0 19 - Reserved Item - ---- 0 20 - Reserved Item - ---- 0 Program 0154 - DSP Gain Setup Use this program to set the DSP gain for the VoIP trunks. Each VoIPU PCB must be programmed. TX Gain is the gain from the PCM highway to the IP network. RX Gain is the gain from the IP network to the PCM highway. Note: If the VoIPU PCB is working, changing this option will cause the PCB to reset. VoIPU Index Number ( 1 5 ). Circuit No Input data Default B1-Channel Tx Gain 0-29 0 B1-Channel Rx Gain 0-29 0 B2-Channel Tx Gain 0-29 0 B2-Channel Rx Gain 0-29 0 B3-Channel Tx Gain 0-29 0 B3-Channel Rx Gain 0-29 0 B4-Channel Tx Gain 0-29 0 B4-Channel Rx Gain 0-29 0 B5-Channel Tx Gain 0-29 0 B5-Channel Rx Gain 0-29 0 B6-Channel Tx Gain 0-29 0 B6-Channel Rx Gain 0-29 0 B7-Channel Tx Gain 0-29 0 B7-Channel Rx Gain 0-29 0 Page 13

Circuit No Input data Default B8-Channel Tx Gain 0-29 0 B8-Channel Rx Gain 0-29 0 Entry Values for Above: 0 = 0, 1 = +1.0, 2 = +2.0, 14 = +14.0, 15 = 0, 16 = -1.0, 17 = -2.0, etc.) Program 0155 - VoIP Timer and Count Setup Program the timer and count value for the VoIP timers. There are three available tables, all using the same entries from the table below. The default settings for each of the tables is the same. These entries do not normally need to be changed from their default setting. These tables are assigned to a VoIP PCB in Program 0153: Item 9, Timer and Counter Setup Table Number. Note: If the VoIPU PCB is working, changing this option will cause the PCB to reset. Item No. Item Input data Default 1 T101 1-254 seconds 5 2 T102 1-254 seconds 50 3 T103 1-254 seconds 50 4 T105 1-254 seconds 50 5 T106 1-254 seconds 5 6 T108 1-254 seconds 50 7 T109 1-254 seconds 50 8 T301 180-254 seconds 180 9 T302 1-254 seconds 15 10 T303 1-254 seconds 4 11 T304 1-254 seconds 15 12 T310 1-254 seconds 10 13 N100 1-254 2 14 Round trip delay timer 0-65535 seconds 0 15 GRQ timer 1-254 seconds 5 16 GRQ count 1-254 2 17 RRQ timer 1-254 seconds 3 18 RRQ count 1-254 2 19 URQ timer 1-254 seconds 3 20 URQ count 1-254 1 21 ARQ timer 1-254 seconds 3 22 ARQ count 1-254 2 23 BRQ timer 1-254 seconds 3 24 BRQ count 1-254 2 25 IRQ timer 1-254 seconds 3 26 IRQ count 1-254 1 27 IRR timer 1-254 seconds 5 28 IRR count 1-254 2 29 DRQ timer 1-254 seconds 3 30 DRQ count 1-254 2 31 LRQ timer 1-254 seconds 3 32 LRQ count 1-254 2 33 RAI timer 1-254 seconds 3 34 RAI count 1-254 2 Page 14

Program 0319 - VoIP Operating Mode The VoIPU PCB work in NT mode. Following program assigns a slot number and operating mode for each VoIP PCB index (1 5). If the VoIPU PCB is working, changing this option will cause the PCB to reset. VOIPU index number (1 5) Cabinet Number 1, Slot Number Options: 4-8 or 0 (No PCB) Cabinet Number 2, Slot Number Options: 12-16 or 0 (No PCB) Cabinet Number 3, Slot Number Options: 20-24 or 0 (No PCB) Mode: 1: NT mode - trunks = 1 TE mode - terminals = 2 (Not Currently Supported) Input Data Default Item 1: Gatekeeper Mode for VoIP Trunk 0 0: Overlap Sending 1: En Bloc Sending Item 2: Reserved 0 Item 3: Reserved 0 Item 4: Reserved 0 Item 5: Reserved 0 Item 6: Reserved 0 Item 7: Reserved 0 Item 8: Reserved 0 When the VoIP PCB is installed, the system will assign 4 trunk ports per VoIP PCB. If there is no slot number in Program 0319, no port numbers will be assigned. Ports are assigned in the order the PCB is installed. Port assignment and the VoIP Index number are independent of each other. Program 0501 - System Numbering If using F-Routing, define the digit(s) to be used to initiate Flexible Routing (9). Program 0603 - Entering Abbreviated Dialing Numbers and Names When using ARS to route calls, enter the digits that will be dialed to reach the destination IP address. These entries must be made in the range specified in Program 0605. Program 0604 - Common Abbreviated Dialing Trunk Group Enter the VoIP trunk group number (01 to 26) for the table number in Program 0928. Program 0605 - Abbreviated Dial Table Area for IP Address To determine the trunk group, the software can search dialed digits in the Abbreviated Dialing Number table. If you want to use this feature, program the range setting below. Entries: Start Address: 0 1999 Length: 0 2000 Default: Start Address: 0 Length: 0 Program 0901 - Basic Trunk Port Setup (Part A), Item 3: CODEC Gain Type VoIP trunks will ignore the setting in this program and will use the settings in Program 0154 - DSP Gain Setup instead. Refer to Program 0901:23 for information on conference call gain settings. Program 0901 - Basic Trunk Port Setup (Part A), Items 14-17, 32-35: Trunk Service Type For each VoIP trunk, set this option to 3 (DID trunk) or 5 (tie line). Use the i-series Software Manual, P/N 92000SWG**), to complete all the required DID or tie line programming. Page 15

Program 0901 - Basic Trunk Port Setup (Part A), Item 23: Unsupervised Conference Call CODEC Gain Type Conference calls may experience an echo. To prevent this, the conference gain settings have been changed for VoIP trunks. If there is a VoIP trunk on a conference circuit, the gain setting will be changed for the conference member who is on a trunk or single line telephone. In these cases, the gain setting in this program will be used for those conference members. If there are two VoIP trunks in the conference, the gain setting from either port will be chosen by the system. The value of the gain should be lower than 0 db, otherwise an echo will occur. Refer to Program 0154 for information on DSP gain settings. Range: Type 1: 1 (0 db transmit and receive gain) Type 2: 42 (-5 db transmit and receive gain) Type 3: 38 (-3 db transmit and receive gain) Type 4: 6 (+3 db transmit and receive gain) Type 5: 10 (+5 db transmit and receive gain) Default: Type 2 setting 42 (-5 db transmit and receive gain) Program 0901 - Basic Trunk Port Setup (Part A), Item 31: Loop Disconnect Supervision For each VoIP trunk, set this option to 1 to enable trunk-to-trunk transfers for the trunk. Program 0905 - Trunk Groups Assign VoIP trunks on each PCB to the same trunk group (1-26). Program 0928 - Destination IP Address Settings This program is used by the F-Route or ARS feature. When not using Gatekeeper, set the IP address and trunk groups. For an IP address, please ask your MIS network administrator. Table Number: 1 96 VoIP Trunk group number: 0 Trunk group MAX (26) IP Address: xxx.xxx.xxx.xxx (xxx: 0 255) Default: All entries are set to 0. Program 2101 - ARS Call Route Options Table For the referring 0928 entry, the trunk group range of F-Route and ARS have been expanded in this program: Trunk Group Entry: 1-26 and 101-126 (101 will point to table 1 in Program 0928, 102 will point to table 2, etc.) Program 2902 - Flexible Numbering Plan Lookup Table With F-Routing, use the entries in this table to associate the numbers identified as type 9 in Program 0501. Program 2903 - Flexible Route Selection Table With F-Routing, when a match is found in Program 2902, the associated data points to an entry in this program (0-48). This is used to select a Route Table (Program 2904) for each day/night mode. Program 2904 - Flexible Routing Tables (Route Treatment) For the referring 0928 entry, the trunk group range for F-Route and ARS have been expanded in this program: Trunk Group Entry: 1-26 and 101-255 however 127 through 254 are not used. (101 will point table 1 in Program 0928, 102 will point to table 2, etc.) Firmware Upgrade The VoIP PCB provides a PCMCIA card slot for firmware upgrades. This slot can only accept an ATA PC card. It should be noted that this card is NOT the same PC card used to update telephone system software. Using the following steps will allow you to use the ATA PC card to update the firmware of the VoIP PCB. 1. Remove the VoIP PCB from the system cabinet. The system should remain powered up. If it has been powered down, wait until the system completely reboots or the VoIP PCB firmware update will not be completed. 2. Insert the PCMCIA card containing the firmware update into the PCMCIA card slot on the PCB. 3. Press and hold the red load switch located on the VoIP PCB and insert the PCB back into the system cabinet. 4. Continue holding in the load switch until the PLT and LED1 LEDs are lit on the VoIP PCB. The system starts the firmware update. LEDs on the PCB will start to light from the highest port to the lowest (8, 7, 6, etc.). 5. Once all LEDs except the CDLED are off, the updating process is complete. The PCB will reboot automatically. 6. Remove the PCMCIA card from the PCB. The CDLED will go out. Page 16

NEC America, Inc., Corporate Networks Group 4 Forest Parkway, Shelton, CT 06484 Tel: 800-365-1928 Fax: 203-926-5458 cng.nec.com Other Important Telephone Numbers Sales:....................................203-926-5450 Customer Service:...........................203-926-5444 Customer Service FAX:.......................203-926-5454 Technical Service:...........................203-925-8801 Discontinued Product Service:..................900-990-2541 Technical Training:...........................203-926-5430 Emergency Technical Service (After Hours)........203-929-7920 (Excludes discontinued products)

*92046INS01* 92046INS01 NEC America, Inc., Corporate Networks Group 4 Forest Parkway, Shelton, CT 06484 TEL: 203-926-5400 FAX: 203-929-0535 cng.nec.com January 2003 Printed in U.S.A.