Bandwidth 1 0
Bandwidth 1 0
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®
INT-2095
Issue 1.0
NEC Corporation of America reserves the right to change the specifications, functions, or features at
any time without notice.
NEC Corporation of America has prepared this document for use by its employees and customers.
The information contained herein is the property of NEC Corporation of America and shall not be
reproduced without prior written approval of NEC Corporation of America.
UNIVERGE is a registered trademark of NEC Corporation. All other brand names and product names
referenced in this document are trademarks or registered trademarks of their respective companies.
Copyright 2009
3.4 SIP System Information Setup (Static and Dynamic Configuration) ..... 1-12
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3.12 IP Trunk (SIP) Calling Party Number Setup for Extensions ..................1-20
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ii Table of Contents
LIST OF FIGURES and tables
___________________________________________________________________________________
Figure 13 IP Trunk (H.323/SIP) Calling Party Number Setup for Trunks ........................... 19
Figure 14 IP Trunk (SIP) Calling Party Number Setup for Extensions ............................... 20
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Bandwidth
1.1 This Guide and Related Documents
For complete details on using SIP trunks with the SV8100, refer to the
SV8100 Networking Manual.
For complete details on using DID features, refer to the DID feature in
the SV8100 Features and Specifications Manual.
1.4 Requirements
A minimum of four IP (SIP) trunks are required due to the NEC Communications
Server infrastructure setup.
The system software for the NEC Communications Server should be version 3.00
or higher.
NEC recommends that the requirements and programming are completed with as
much information as possible before scheduling an activation appointment with
Bandwidth.
1.5 Limitations
This section provides information to NEC’s solution providers and NEC Associates for
configuring an NEC UNIVERGE SV8100 to connect to a Bandwidth SIP Trunk service
provider, utilizing a DYNAMIC and STATIC configuration.
2.1 Prerequisites
Before you configure the UNIVERGE SV8100, you must have the following
information available.
WAN Side:
WAN IP Address:
LAN Side:
VLAN ID:
PBX Information:
Model:
Firmware Version:
Number of Users:
Notes:
When using Bandwidth as your SIP trunking service provider, the following programs
must be changed for SIP trunking service.
Values shown are for example purposes only. Your actual IP values will be
determined by your local LAN administrator.
If a router or firewall is placed between the SIP Trunk Provider and SV8100,
you must also set the following programs:
The SV8100 must be reset in order for the change to take effect.
Values shown are for example purposes only. Your actual IP values will be
determined by your local LAN administrator.
Port Forwarding:
The Router will require port forwarding rules to be configured.
The ports used in Programs 84-26-02 and 84-26-03 must be forwarded to the
IP address entered in Program 84-26-01.
The RTP/RTCP ports are forwarded to avoid possible one-way conversation
which might occur on inbound calls. When forwarding the ports, the range for
each gateway must be set. The number of gateways to forward will depend on the
size of the IPLA.
Gateway 1 will require ports 10020-10051 forwarded.
Gateway 2 will require ports 10052-10083 forwarded.
Gateway 3 will require ports 10084-100115 forwarded.
Gateway 4 will require ports 10116-10147 forwarded.
Gateway 5 will require ports 10148-10179 forwarded.
Gateway 6 will require ports 10180-10211 forwarded.
Gateway 7 will require ports 10212-10243 forwarded.
Gateway 8 will require ports 10244-10275 forwarded.
5060 Yes No
10020 Yes No
10021 Yes No
10052 Yes No
10053 Yes No
10084 Yes No
10085 Yes No
10116 Yes No
10117 Yes No
IPLA32 2
IPLA64 4
IPLA128 8
udp;143.101.120.218/255.255.255.0-10020>10.10.3.20-10020
udp;143.101.120.218/255.255.255.0-10021>10.10.3.20-10021
udp;143.101.120.218/255.255.255.0-10052>10.10.3.21-10052
udp;143.101.120.218/255.255.255.0-10053>10.10.3.21-10053
udp;143.101.120.218/255.255.255.0-10084>10.10.3.22-10084
udp;143.101.120.218/255.255.255.0-10085>10.10.3.22-10085
udp;143.101.120.218/255.255.255.0-10116>10.10.3.23-10116
udp;143.101.120.218/255.255.255.0-10117>10.10.3.23-10117
udp;143.101.120.218/255.255.255.0-5060>10.10.3.10-5060
Values shown are for example purposes only. Your actual values will be
determined by your implementation team.
Values shown are for example purposes only. Your actual values will be
determined by your implementation team.
10-29-08 : SIP Server Information Setup – SIP Proxy Setup – DNS Mode
Set the DNS Mode to 1, when the SIP carrier provides a domain name.
10-29-09 : SIP Server Information Setup – SIP Proxy Setup – DNS IP Address
This information should be provided by your SIP service provider.
The DNS IP Address should be any valid Domain Name Server either SIP provided or
within your network.
This field applies for those SIP service providers supporting Static
Configuration.
Values shown are for example purposes only. Your actual values will be
determined by your implementation team.
Values shown are for example purposes only. Your actual values will be
determined by your implementation team.
Caller ID - In the Invite message there are two fields that can have caller ID. One
field is the “SIP From Address” and the other field is “SIP Display Info”. If both of
these fields are left blank the call will not complete.
Values shown are for example purposes only. Your actual values will be
determined by your implementation team.
Values shown are for example purposes only. Your actual values will be
determined by your implementation team.
This setting has the highest priority. This program is used to assign the Calling
Party Number for each extension (Entries: 1~0, *, #). The assigned number is
sent to the SIP Trunking Service Provider when the caller places an outgoing call.
If the Calling Party Number is assigned by both Program 21-17 and 21-18/21-19,
then the system uses the data in Program 21-18/21-19. Do not use Program
21-13 for SIP. This entry must be a 10-digit DID associated with the SIP Trunking
Service Provider Account. DID numbers are provided by your SIP Trunking
Service Provider Coordinator.
Values shown are for example purposes only. Your actual values will be
determined by your implementation team.
Values shown are for example purposes only. Your actual values will be
determined by your implementation team.
84-13-32 : SIP Trunk CODEC Information Basic Setup – DTMF Relay Mode
Set the DTMF setup to 1 (RFC2833).
Values shown are for example purposes only. Your actual values will be
determined by your implementation team.
The SV8100 must be reset in order for the change to take effect.
The SV8100 must be reset in order for the change to take effect.
Values shown are for example purposes only. Your actual values will be
determined by your implementation team.
Changes within this program require the SV8100 be reset in order for the
change to take effect.
To confirm that the system is correctly set, perform the following tests:
If you run into an issue with any of these tests, refer to Table 4 Troubleshooting Guide.
Test an outgoing call to a local number. Check for ringback, 2-way audio and quality.
1. Test an outgoing call to a long distance number. Check for ringback, 2-way
audio and quality.
4. Test multiple call concurrences on outgoing calls. Setup multiple calls to PSTN.
8. Test an incoming call to an internal DID. Check for ringback, 2-way audio and
quality.
Missing Parts of Words Packet Loss or Latency on WAN Check with Internet Access Provider
The following are screen shots taken from an active NEC InRouter while interfaced
with BandTel SIP service. They are provided as a guide; your situation could be
different from what is shown. The IP Address shown will of course not apply as you
will need the IP Address assigned to your unique situation.
5.1 Network
5.2 Firewall
5.3 NAT
5.6 Rules
Rules are used to forward and/or modify incoming and outgoing calls. There are
three types of rules:
Inbound – from server to trunking device
Outbound – from trunking device to server
Redirect – from local telephone to trunking device (w/o routing to server)
Outbound rules can match against and/or modify either the calling or called
number. Inbound and redirect rules operate on the called number only. Stripped
and added digits always apply to the left-most digits of the DID.
These header transformation rules are applied to all SIP trunking devices. They
define how specified SIP headers should be transformed when forwarding to the
SIP Server.