Quintum | Tenor AS | TN037 - Quintum Tenor AS Gateway Installation & Configuration

NetVanta Unified Communications Technical Note
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Installing and Configuring the Quintum
Tenor AS Gateway
Introduction
The Quintum Tenor AS is a 2-4 port analog gateway used in NetVanta Unified Communications Server
installations to provide a gateway between internal (SIP) phone calls and the outside phone network
(PSTN). Voice communications from an internal phone have voice over IP (VoIP) signals converted into
traditional analog voice, which is transmitted over the PSTN.
A gateway works in conjunction with the UC server’s SIP Proxy and SIP. All telephony services are
provided through the mutual cooperation of SIP gateways, SIP telephones, SIP proxy and the Core
Application Service.
The following diagram illustrates the UC server’s SIP architecture and its relationship with other
components in a typical customer network.
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Supported Features
Feature Name
Supported
Accept Incoming Calls

Accept Outgoing Calls

Trunk-to-trunk connect

Calling Party Name

Calling Party Number

Answer Supervision

Disconnect detection

DTMF Tone Support (RFC2833
Compliant)

Conferencing with SIP Endpoints

Direct Inward Dialing

System Music on Hold Support

Outgoing Fax Support

Incoming Fax Support

Unified Communication Features Supported by Gateway
Active Message Delivery

Paging Notification

Transfer—Assisted/Supervised

Transfer—Blind

Multiple SIP Proxy Support
 *Available with survivability
option
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Interoperability Software Versions
The following gateway version was tested for interoperability:
System Description: Quintum Tenor AS
Hardware Version: P106-02-00
Firmware Version: P106-12-00
Overview of Procedure
To provide its functionality, the Quintum Tenor AS must be connected to the internal LAN (a 100 Mbps
connection is recommended) and from 1-4 PSTN analog phone lines.
The Quintum Tenor AS is primarily configured using a java configuration program. The program must
be installed to configure and manage the gateway.
The basic steps for installation and configuration are:
1. Unpack the Quintum Tenor AS.
2. Mount the Quintum Tenor AS.
3. Connect cables.
4. Power up the Quintum Tenor AS.
5. Set a DHCP IP address reservation for the Quintum Tenor AS based on its MAC address.
6. Run the initial configuration wizard.
7. Configure the UC server to use the Quintum Tenor AS.
Note: Please see the instructions provided by Quintum for steps 1 to 4, and for information about running
and configuring the gateway.
The rest of this document provides instructions for steps 5 to 7, which allow you to configure the
Quintum Tenor AS for operation with the UC server.
Address Reservation
By default, the gateway is configured to use an IP address assigned by DHCP. The gateway can also be
configured to use a static IP address. For routing calls out from the UC server, the Quintum Tenor AS
must have an IP address that does not change.
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Initial Configuration
Installing Tenor Configuration Manager
To begin configuration of the Quintum gateway, you must first install the Tenor Configuration Manager.
You can either get it from the CD included with the gateway or at the Quintum support website
(http://www.quintum.com/support).
After you have installed and run the Tenor Configuration Manager, the following screen appears.
Adding the Gateway
If your PC is running on the same subnet as the gateway, the gateway can be added automatically. If your
PC is running on a different subnet than the gateway, the gateway must be manually added.
To add the gateway automatically
1. Select Discover to automatically detect the gateway.
2. When the wizard finds the gateway, select Connect.
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To add the gateway manually
1. Select Add.
The following screen appears.
2. Enter the IP Address of the gateway.
3. Enter admin as the username and password.
4. Select OK.
5. Select Connect on the Address Book screen.
Running the Configuration Wizard
After you connect, a wizard opens to set up the initial configuration of the gateway.
1. Select Next.
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2. On the screen below, you have the option to choose how your gateway obtains its IP Address and
network settings. A static IP address is recommended for a gateway.
3. You can specify whether you want to obtain DNS server addresses automatically or if you want to
manually configure them. If you are using DHCP, you can automatically obtain the DNS server
addresses; otherwise you must manually configure them. Select Next to continue.
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4. The first task is complete. Select Next to continue.
5. The Dial Plan Configuration screen allows you to set up the dialing plan. Choose None from the
Dial Plan Country list. Currently the dial plan rules do not work and will result in outgoing calls not
working. When finished, select Next to continue.
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6. On the Phone Port Configuration screen, choose the method for Disconnect Generation and Caller
ID Generation. These settings depend on your carrier and location. When finished, select Next to
continue.
7. On the Phone Port Configuration screen, you can map DID numbers to individual channels on the
gateway. When finished, select Next to continue.
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8. On the Multipath Configuration screen, you can choose for all calls to automatically pass through
between the phone side and lines. You do not need to enable Pass Through calls, so you can select
No.
9. On the Add Bypass Number screen, select Done, and then select Next.
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10. On the Line Port Configuration screen, choose the method for Disconnect Detection and Caller ID
Detection. These settings depend on your carrier and location. Select Next to continue.
11. On the VoIP Routing Configuration screen, for integration with the UC server, choose SIP only.
Select Next to continue.
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12. In the SIP Server Information section, change the Primary SIP Server to the IP address of your UC
server. When finished, select Next to continue.
13. In the Add SIP User Information section, the wizard requires you to enter a User ID and Password.
Enter 9999 for both fields. Select Done and then Next to continue.
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14. In the Idle Channel Configuration screen, select Yes if you are not using the maximum number of
phone ports on your gateway. Select Next to continue.
15. If you selected Yes on the previous screen, you will be presented with the screen below. Clear the
checkboxes of the ports that are not used. Select Next to continue.
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16. In the Idle Channel Configuration screen, select Yes if you are not using the maximum number of
PSTN ports on your gateway. Select Next to continue.
17. If you selected Yes on the previous screen, you will be presented with the screen below. Clear the
checkboxes of the ports that are not used. Select Next to continue.
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18. On the Configuration Summary screen, check the settings to make sure everything is correct. You
can go back and make changes if necessary. When finished, select Accept to continue.
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19. After the initial configuration is done, reboot the Quintum gateway and navigate to the Line Port
Configuration tab.On that tab, you can set up numbers that are allowed go through to the PSTN from
the SIP side. A hopoff number would usually contain the first few digits of a PSTN number based on
your location. For example, to allow local calls you would add an entry with 613 as the number
pattern and replacement number where 613 is your local area code. If you want to allow
international numbers for North America then you would use 011 as the number pattern and
replacement number. When finished, select Confirm/OK.
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20. Select the Advanced Explore tab, and navigate to VoIP Configuration > Voice Codecs > Voice
Codec-1. Set Voice Codec to G.711 Mu-law 64 kb. Select Confirm/OK.
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21. Navigate to Circuit Configuration > Trunk Circuit Routing Groups > Trunk Circuit Routing
Group-line. Under the Advanced tab and in the Forced Routing Number box, enter the autoattendant identity. Typically, this is set to 10000.
22. Under the Trunk ID/Caller ID tab and in the Trunk ID Delivery list, choose Calling Party
Number.
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23. Navigate to VoIP Configuration > SIP Signaling Groups > SIP Signaling Group-1. Under the
Advanced tab, make sure that Nortel is selected in the SIP Info Format list.
24. Navigate to VoIP Configuration > IP Routing Groups > IP Routing Group-default. Under the
ANI tab and in the Relay Calling Name list, choose Relay CNAM in Invite. Select Confirm/OK.
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25. To complete the changes, select Confirm/OK and then select the submit changes button.
Enabling CNG Tone Detection for Faxing
By default, a Quintum gateway will not detect CNG tones used for faxing unless the call is directed at a
fax service. In order to receive faxes when a call is answered by a standard service (not a fax service), you
must create a file and upload it to the gateway via FTP.
To enable CNG detection
1. Open notepad or another text editor.
2. Put in the following line: enableCNGdetection 1
3. Save the file as var_config.cfg.
4. From your Windows PC select Start > All Programs > Accessories > Command Prompt. The
Command Prompt window is displayed.
5. Use the CD command to change to the directory on your PC in which you saved the
var_config.cfg file.
6. Type ftp followed by the IP address of the unit. Press Enter.
7. Login with the username and password. The default for both is admin.
8. Use the CD command to change to the cfg directory (this is the directory on the Tenor into which
you will copy the var_config.cfg file). Depending upon the product type and software revision,
the directory structure you see in your Tenor VoIP device may be different.
9. Type bin <Enter>.
10. Type put var_config.cfg <Enter>
11. Restart the gateway from the Tenor Configuration Manager in Tools > Reboot Tenor.
Configuring the UC Server
After you add the gateway to your network, the UC server must be configured to handle incoming and
outgoing phone calls. For outgoing calls you must add: a SIP gateway, a dial plan entry to route calls out
through the gateway, and a toll restriction entry to allow those calls. For incoming calls you must add a
UC server identity that can answer incoming calls from the gateway. These instructions are for release 4.1
of the UC server.
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.
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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 selected service.
Adding a SIP Gateway
1. Select Gateways.
2. Right-click the right panel and select New Gateway.
3. Choose Public Switched Telephone Network (PSTN) from the gateway list.
4. In the Host name field, enter the IP address of the gateway.
5. Enter a descriptive name for the gateway.
6. Save.
Configuring the Dial Plan
Incoming calls from the PSTN are already configured by having incoming calls routed to the 10000
Trunk identity. An entry or entries must be entered in the Dial Plan for outgoing calls to the PSTN.
1. Go to Communication Service > UC Server > Routing.
2. There are many possibilities here. 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 gateway. For example:
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Configuring Toll Restrictions
Configure the toll restrictions to match the requirements of your organization. Consult the NetVanta
Unified Communications Server Administrator Guide, available online at http://kb.adtran.com, for the
correct use of regular expressions in the toll restrictions to enforce corporate dialing policy. It is explained
in detail in the “Routing and Restricting Calls > Allowing and Restricting Long Distance and Other Calls
> Restricting long distance calls” section.
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Glossary of Features
Accept Incoming Calls
This feature allows the gateway to answer an incoming call from the PSTN. The gateway then makes a
SIP call to extension 10000.
Accept Outgoing Calls
An outgoing SIP call from the UC server results in an outgoing PSTN call.
Active Message Delivery
The gateway must support the UC server calling out to the PSTN to deliver voice messages.
Answer Supervision
The gateway must detect that a call has been answered. There are a number of techniques used for this,
including loop start, battery reversal and voice detection.
Calling Party Name
The gateway detects the calling party name on an incoming PSTN call and provides that name to the UC
server.
Conferencing with SIP Endpoints
The gateway needs to support conferencing between itself and other SIP endpoints.
Direct Inward Dialing
Calls incoming from the PSTN must be automatically routed to the UC server for auto attendant
functionality.
Disconnect Detection
The gateway must detect that a call has been dropped. There are a number of techniques used for this,
including loop start, battery reversal and no voice detection.
DTMF Tone Support (RFC2833 Compliant)
Calls incoming from the PSTN to the UC server are usually handled by an auto attendant. Feature
operation is implemented using DTMF tones from telephones. These tones must be sent to the UC server
as SIP packets via RFC2833.
Incoming Fax Support
The UC server supports the transmission of faxes to standard fax machines. The gateway must support
T.38 fax transport to provide this capability. Additionally, the gateway should support CNG tones so that
an incoming PSTN fax call can be distinguished from a voice call and handled appropriately.
Multiple SIP Proxy Support
In high reliability applications, if the main UC server is not available the gateway routes incoming PSTN
calls to an alternative SIP Proxy.
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Outgoing Fax Support
The UC server supports the transmission of faxes to standard fax machines. The gateway must support
T.38 fax transport to provide this capability.
Paging Notification
The gateway must support the UC server calling out to the PSTN to deliver pages.
System Music on Hold Support
The UC server supports music on hold. When PSTN callers are on hold they hear music, if that feature is
enabled on the system.
Transfer—Assisted/Supervised
After a call is established between an outside PSTN call and an internal SIP device, the gateway must
allow a supervised transfer to another SIP device.
Transfer—Blind
After a call is established between an outside PSTN call and an internal SIP device, the gateway must
allow a blind transfer to another SIP device.
Trunk-to-trunk connect
This feature allows an established call through the gateway, which can be extended back out the gateway
on another PSTN trunk.
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