NetVanta Unified Communications Technical Note

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Installing and Configuring AudioCodes

Mediant 1000

1 Introduction

The AudioCodes Mediant 1000 (M1K) is a T1 digital gateway that can be used in NetVanta Unified
Communications Server installations to provide a bridge between internal (SIP) phone calls and the phone
network (PSTN). The M1K supports 1, 2 or 4 T1 spans. It bridges SIP VoIP phones on the Local Area
Network (LAN) and the traditional TDM voice network (PSTN).
A gateway works in conjunction with the SIP Proxy and Registrar that are part of the UC server. All
telephony services are provided through the mutual co-operation of the SIP Gateway, SIP Telephones,
SIP Proxy and the Core Application Services.
The following diagram illustrates the UC server SIP Architecture and its relationship with other
components in a typical customer network.

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This document is a quick start guide to integrating the gateway with UC Serer. It is not a replacement for
the manufacturer’s documentation, and specific customer requirements might fall outside the guidelines
of this document.
It is recommended that skilled technicians obtain training directly from the manufacturer for gateway
configuration and deployment.

2 Overview of Procedure
To provide its functionality, the M1K must be connected to the internal LAN (a 100 Mbps connection is
recommended) and 1, 2 or 4 T1 digital spans.
The M1K has two configuration methods: Web page or configuration file. This document describes
configuration using the gateway’s Web page. This has the advantage of providing familiarity with the We
page method, which will most likely be used later for making minor changes in the gateway
configuration. It has the disadvantage that an inadvertent change in one of the many parameters might
cause problems. For this reason it is recommended that only the changes described below are made
to a factory reset gateway. After the gateway is operational, further changes may be made as required.
The basic steps for installation and configuration are:
1. Unpack the M1K.
2. Mount the M1K.
3. Connect cables.
4. Set the IP address and subnet mask of the gateway through the serial port on M1K.
5. Access the M1K Web page.
6. Configure the gateway.
7. Save the configuration and reboot the M1K.
8. Back up the configuration.
Steps 1-3 are standard for any gateway. Please follow the instructions provided by AudioCodes for the
gateway. AudioCodes provides comprehensive information on the use and configuration of the gateway.
This document is intended to be a companion document to the AudioCodes documents, which should be
used while configuring the gateway.
The rest of this section details Steps 4 and 8 to configure the M1K for operation with the UC server.

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3 Configuring the Gateway
3.1 Software Level
This document is based on the gateway software from the last interoperability testing conducted by
ADTRAN. If the software revision does not match, contact the manufacturer’s website for the version
that is shown below. Failure to do so might result in unexpected behavior.
To confirm the software revision for your gateway, navigate to Status and Diagnostics -> Device
Information and make sure that the gateway software matches the information below before proceeding.
Versions Version ID: 5.20A.047.003

3.2 Setting the IP Address and Subnet Mask

It is important that the M1K have a LAN IP address that does not change. This can be set using a static IP
address and subnet mask compatible with the on-site LAN. Please follow the instructions provided by
AudioCodes. A summary of the procedure is:
1. Connect the M1K and a PC on the same LAN subnet.
2. Use or install from the AudioCodes CD a bootp application.
3. Power up the M1K.
4. Edit the IP address, subnet mask, and gateway information in the bootp information window. See
Figure 1 below.

5. Now access the gateway’s Web page using your browser and the M1K’s new IP address. The
default login information is the following:
Username: Admin
Password: Admin

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3.3 Configuring SIP
When the gateway receives a call from the PSTN it must know where to send that call. In the reverse
direction, the gateway must accept SIP calls from the UC server and direct those calls out to the PSTN.
This is generally configured by providing SIP and Dial Plan configuration information.
The UC server IP address or name within the enterprise domain is the SIP local domain for the gateway.
Standard UC server configurations for gateways do not require that the gateway register on the UC server
as a SIP identity.
Calls between PSTN devices and services on the UC server may make use of DTMF tones, for example,
voice mail and auto attendant functions. The DTMF digits must be transported outside the voice stream to
the UC server. This is done by enabling DTMF Transport using rfc2833.
Similarly, faxes that are sent or received by the UC server must be supported by transmitting the fax
information outside the TDM voice path. This is implemented using T.38 fax support. This must be
enabled on the gateway by doing the following:
1. Navigate to Quick Setup.
2. Leave the Gateway Name blank.
3. Make sure that Working with Proxy is set to Yes.
4. In Proxy IP Address enter the IP address of the UC server.
5. Leave the Proxy Name blank.
6. Click Reset and when prompted save the information.
See Figure 2 below.

Figure 2
7. Go to Protocol Management -> Routing Tables -> Tel to IP Routing.

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8. Enter a row that has Dest. IP Address set to the the UC server IP address.
9. Click Submit.
See Figure 3 below.

Figure 3
10. Go to Advanced Configuration -> Media Settings -> Voice Settings and make sure that DTMF
Transport Type is set to RFC2833 Relay DTMF. See Figure 4 below.

Figure 4
11. Go to Advanced Configuration -> Media Settings -> Fax/Modem/CID Settings and make sure
that:
Fax Transport Mode is T.38 Relay.
V.21 Modem Transport Type is set to Disable.

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See Figure 5 below.

Figure 5
12. Go to Protocol Management -> Protocol Definition -> Coders and make sure that G.711U-law and
G.729 codecs are chosen (for North America).

Figure 6
13. Go to Protocol Management -> Protocol Definition -> General and ensure that Enable Early
Media is set to Enable.

Figure 7

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3.4 Dial Plan
Routing or dial plan entries must be configured to route calls in from the PSTN to the UC server, and out
from the UC server to the PSTN.

3.4.1 Incoming
There must also be an answering point defined on the UC server for incoming PSTN calls. A standard
configuration has a UC server trunk identity of 10000. Enter a general routing entry on the gateway so
that all incoming calls are sent to 10000 on the UC server.
When the call is presented from the PSTN, the carrier provides the called party information. Often this is
not the full 10-digit number, but is the last 4 digits of the number being called. In either case that number
may be used to route the call to a destination.
Go to Protocol Management -> Manipulation Tables -> Tel -> IP Destination Numbers and:
1. Add an entry with:
Destination Prefix set to *.
Source Prefix set to *.
Number of stripped Digits set to the number of incoming digits given by the carrier.
Prefix (Suffix) to Add set to the identity that should answer the call (for example,
10000).
2. Click Submit.
See Figure 8 below.

3.4.2 Incoming DID Call Routing

The entry routes all incoming PSTN calls to identity 10000 no matter what number is called. If there are
multiple incoming DIDs, more routing entries must be added.

Figure 8 – in this example, the main DID is 9698 and it is routed to the trunk identity 10000. There are a
number of other incoming DIDs. An example is 2180, which is routed to the 707 identity. The first rule
that matches is the one that is used, so it is important to have any catch-all rules (for example, a
destination of *) at the bottom of the tables.

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3.4.3 Outgoing
A gateway may have more physical and logical trunks than are being used at a site. When an outgoing
call is made to the PSTN it is important that the gateway not choose a trunk that is not there or not in
service. Enable only those physical interfaces and trunks that are actually in use.
If an operational trunk is busy, the gateway must ‘hunt’ for an unoccupied trunk to complete the call.
Configure a trunk group that specifies which trunks are to be used for outgoing calls and what the hunting
algorithm should be, for example, ascending or descending.
1. Go to Protocol Management -> Trunk Group
2. Create a Trunk Group ID (for example, 1) that includes all trunks and channels used for
outgoing calls.
The phone number assignment is arbitrary. A number should be assigned but in general will not be used
for most installations.

Figure 9 – In this example there are two trunk groups, 1 and 2. Trunk group ID 1 is using the first T1
span and has channels 1-8 enabled. If a call is presented to that group, those channels are used.
3. Go to Protocol Management -> Trunk Groups Settings
4. Choose the type of outgoing trunk behavior required, for example, Descending.

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 Figure 10
5. Go to Protocol Management -> Routing Tables -> IP to Hunt Group Routing.
6. Add an entry that maps all outgoing calls to the trunk group defined.

Figure 11 – the last entry is the default route that takes any number and sends it to Trunk Group ID 1.

3.4.4 Outgoing DID Presentation

When an outgoing call to the PSTN is made over an analog trunk, the caller ID is generated by the carrier.
There is no ability to change the ID on a call-by-call basis. A PRI allows the calling party to specify the
caller ID.

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 1. Go to Protocol Management -> Manipulation Tables -> IP -> Tel Source Numbers.
2. If an enterprise-wide caller ID is required, add an entry that maps any source and destination
number to the phone number of the enterprise.
3. If other outgoing phones are desired for some phone number, add table entries for those above the
catch-all entry.

Figure 12 – The last entry is the catch-all that takes any destination and source and presents 6135999698
as the calling number. The second entry, as an example, means that anytime an extension 2220-2229 calls
any number, the presented phone number is 6132712171.

3.5 DSL
The DSL / PRI settings are site- and carrier-specific. This section covers the most likely settings for North
America.
1. Go to Advanced Configuration -> PSTN Settings -> Trunk Settings.
2. Select the trunk to configure by clicking the Trunk Status icon at the top of the page. If the trunk
cannot be selected, it is because it is active. If that is the case, go to the bottom of the page and
stop the trunk.

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Figure 13
3. Protocol Type should be set as defined by your carrier. DMS, ATT and NI are the most common
settings.
4. Clock Master should be set as defined by your carrier. Recovered is the most common.
5. Line Code should be set as defined by your carrier. B8ZS is the most common.
6. Framing Method should be set as defined by your carrier. T1 Framing ESF CRC6 is the most
common.
7. ISDN Termination Side should be set as defined by your carrier. User side is the most common.
8. Local ISDN Ringback Tone Source should be set as defined by your carrier. Gateway is the
most common.
9. Apply Trunk Settings.
10. Click Submit.

3.6 Backing up the Configuration

1. Go to Maintenance. Click Burn under Save Configuration to save all the changes.

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 2. Go to Advanced Configuration -> Configuration File and get the .ini file. Store the
configuration file in a safe place.

4 Configuring the UC Server

After the gateway has been added to your network, the UC server must be configured to handle incoming
and outgoing phone calls.
These instructions are for Release 4.1 of the UC server. Start the UC server admin client.

4.1 Adding a Trunk Identity

1. Go to Identities.
2. Right-click the right panel and select New Identity.
3. In the first page of the Wizard, select an Attendant identity. Make sure that the Identity is
associated with the Admin profile.
4. On the following page, enter a descriptive name and enter 10000 for the Address (assuming a
standard configuration). Make sure that Default Trunk Service is the service to be run.

4.2 Adding a SIP Gateway

1. Click Gateways.
2. Right-click the right-hand panel and select New Gateway.
3. Choose Public Switched Telephone Network (PSTN) from the list of gateways.
4. In the Host name field, enter the IP address of the gateway.
5. Enter a descriptive name for the gateway.
6. Click Save.

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4.3 Dial Plan
Incoming calls from the PSTN are already configured by routing incoming calls to the 10000 Trunk
identity. An entry or entries must be entered in the Dial Plan for outgoing calls to the PSTN. There are
many possibilities here, and the procedure below presents one possibility.
1. Go to Communication Service -> UC Server -> Routing.
2. If regular PSTN calls are to be routed out the gateway, add or modify an entry where the
Original Digits are [0-9]{7,} and select the Vega gateway. For example:

4.4 Toll Restrictions

Configure the toll restrictions to match the requirements of your organization. Consult the UC Server
Administration manual for the correct usage of regular expressions in the toll restrictions to enforce
corporate dialing policy. Toll restrictions are explained in detail in the Managing PBX Configuration
Categories > Routing—Toll Restrictions section of the NetVanta Unified Communications Server
Administrator Guide, available online at http://kb.adtran.com.

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