Sonexis Release 10 Specs

D ATA S H E E T ConferenceManager 10.0
Sonexis ConferenceManager is an in-house audio conferencing bridge that eliminates the costly
pay-as-you-go fees of subscription-based services, while setting new standards for security and
ease of use. Designed to work within existing voice and data networks — and available with a fully
integrated Web conferencing option — ConferenceManager has been deployed by hundreds of
leading organizations with minimal cost, effort, or administration. In fact, Sonexis has delivered
more audio conferencing bridges than any other vendor serving the small and medium-sized
enterprise.

DISCOVER THE BENEFITS:

An Irresistible Economic Story The financial picture is
clear. Sonexis ConferenceManager provides a rapid,
compelling payback. Your service provider just can’t
compete. A minimal one-time investment and your will be scheduling, launching, and recording confer-
organization begins saving immediately regardless of ences; collaborating over files in real-time; even
how your conferencing usage scales. Watch your inviting new conference attendees on the fly.
savings—not your costs—skyrocket.
ConferenceManager makes it very easy for everyone.
Better still, you pay only for what you need. With our Familiar applications (like Microsoft Outlook, Lotus
unique architecture you're never more than a license Notes, and the Web browser) provide the interfaces
key away from increasing users, adding Web func- that make ConferenceManager accommodating for
tionality, or changing from one protocol to another. your end-user; while simple Web-based administra-
Simply put, it's the best thing to happen to conferenc- tion provides the flexibility you need to meet your
ing. organization’s needs.
Unprecedented Ease of Use and Administration The Sensible Protection for Your Business-Critical
ability to adapt ConferenceManager to your changing Information ConferenceManager is an outstanding
needs through a simple software upgrade is just the audio conferencing bridge with integrated Web
beginning of ConferenceManager’s unmatched ease conferencing that sits on your network under your
of use. Whether you’re an end-user or the system management. ConferenceManager’s smart, embed-
administrator, you’ll be surprised that conferencing ded security features keep your information safe from
can be so easy. Simply plug ConferenceManager into outside intrusion. You maintain control of critical
your voice (PSTN) or data network (VoIP) and within business information and the security procedures
minutes your organization that protect it.

ConferenceManager’s powerful features give you fast, full-function v ActiveTalker™ highlights who’s speaking—
audio conferencing that is cost-effective, easy to use, and secure. or who’s got the noisy line

y Host can easily label
sub-conference rooms
and navigate between them

u
Host can easily identify
participants by name
w x
Visual indicator shows Web Participants can signal: Host can divide participants
conference participation into (up to) ten private
When they’re ready to return
Administrators can easily to the main conference; or sub-conferences
manage users through
When they have a question in
listen-only “lecture” mode

CONFERENCEMANAGER 10.0
Audio Conferencing
• Supports 12 to 576 ports PSTN (PRI, CAS) • Integrated Web conferencing (optional)
• Supports 12 to 600 ports VoIP (SIP, H.323) • Supports 5 to 600 seats
• Scheduled or ad-hoc (reservation-less) conferences • Scheduled and ad-hoc (reservation-less) conferences
• Conference scheduling and management through • Cross platform and no download for participation
Microsoft Outlook, Lotus Notes, or Web interface • Scheduling and mgmt via Outlook, Notes, or Web
• Integrated, Web-based host control panel • Pre-defined and saved conference rooms
• Real-time speaker identification (ActiveTalker™) • Application- and document-sharing
• Emergency Management and Blast Dialout • Integrated with audio console
• Supports up to ten sub-conferences off the main • Interactive whiteboard, annotation tools, and notepad
conference
• Follow-me Web browsing
• Advanced IVR for greetings and help menu
• Text chat, Q&A, polling, and voting
• Dial-in or dial-out
• Automatic e-mail invitations, Quick invite System Information
• Multiple conference announcement options • OPERATING SYSTEM: Windows 2008 Server
(tone, name, none)
• PROCESSORS: Intel® 5504TM Quad Core 2.0GHz
• Audio recording
• MEMORY: 4GB Fully Buffered DDR3 RAM
• Call detail records
• STORAGE: 300GB HD
• JITC Certified
• REDUNDANCY: Dual HD (RAID 1) and power (optional)
• ADMINISTRATION: Web, attached KVM, or terminal services
• NETWORKING: Dual Gigabit Ethernet
(10BASE-T, 100BASE-TX, and 1000BASE-T, RJ45 output)
300 Seco Road
Monroeville, Pennsylvania 15146 • EXPANSION: 4 PCI expansion slots
888-476-6394 or 888-4SONEXIS • MOUNTING SYSTEMS: 2U 19”
www.sonexis.com v10.0

KEY FEATURES
- Intel® 2600 Chassis - SR2600URLX
- Intel® 5504 Quad Core™ 2.0GHz processor
- 3GB Fully Buffered DDR3 RAM
- Dual Intel® Pro/1000 EB Ethernet ports
- 2 Riser Slots With Intel® Adaptive Slot Technology
- Cheetah 300GB SAS Hard Drive
- RAID 1 and redundant power (optional)

MOTHERBOARD
• Intel® S5520UR Server Board

PROCESSOR/CACHE INPUT/OUTPUT
CPU • Intel® 5504 Quad Core 2.0 GHz with USB • 2 USB ports
4MB Intel® smart cache Serial Ports • 1 RJ45 Serial B Port
System Bus • 1066MHz/1333MHz system bus Keyboard • Stacked PS/2 keyboard and mouse
/Mouse ports

SYSTEM MEMORY SYSTEM BIOS
Memory • Twelve DIMM sockets BIOS Type • Intel® 28F320C3B 4MB Flash ROM
Capacity • 3GB Fully Buffered DDR3 RAM • Five 3.5” IDE drive bays
• (w/o CMA) 27.75” (704.8mm)
• Dual-channel memory bus (memory • One Optional Hard Drive or Tape
must be populated in pairs) Drive Bay
Memory Type • DDR3 Fully Buffered DIMMs • IDE Slim Optical Drive Bay

ON-BOARD DEVICES DIMENSIONS
Chipset • Intel® E5520 Chipset Height • 3.44” (87.30mm)
IPMI • IPM Interface v.2.0 Width • (w/o Rails) 16.93” (430mm)
• Support for Intel® Remote • (w Rails) 17.77” (451.3mm)
Management Module 3 and Local Depth • (w/o CMA) 27.75” (704.8mm)
Control Panel • (w CMA) 33.0” (838.2mm)
Network • Dual Intel® Gigabit Ethernet Weight • 65 lbs (29.5kg)
Controllers • Supports 10BASE-T, 100BASE-TX, Color • Black
and 1000BASE-T, RJ45 output
Graphics • ATI with 64MB of video memory

EXPANSION SLOTS POWER SUPPLY
• 2 Riser Slots - Intel® Adaptive Slot • 750W AC power supply
• 4 PCI Express buses (2x8 or 4x4) AC Voltage • 100 - 220V, 50/60Hz, 12 amp
• 1 PCI-X bus DC Output • 12V + 5V + 3.3V < 840W
Draw • 2 amps steady, 6.8 max, per supply
Environ. • 750 BTUs steady, per supply

Sonexis Technology, Inc. | 300 Seco Road | Monroeville, PA 15146 | 1-866-2SONEXIS (1-866-276-6394)
marketing@sonexis.com | www.sonexis.com

SONEXIS PRE-IMPLEMENTATION SITE REQUIREMENTS
TDM Systems (T1 and/or PRI connections):

Direct TELCO connection:

1. Order the necessary T1/CAS or T1/PRI circuits.
2. Obtain a CSU/DSU (Channel Service Unit/Data Service Unit).
Sonexis ConferenceManager supports the following protocols, tell your provider you need a circuit
configuration that matches one of the types below:

• CAS - Frame Type: ESF/B8ZS or D4/AMI
(E&M WINK ONLY)
• PRI - Frame Type: ESF/B8ZS or D4/AMI
Switch Types: ATT 4ESS, ATT 5ESS, DMS100, DMS250, and NI2
NO support for Shared D-channels across multiple PRIs (NFAS)

Behind the PBX: (Your PBX T1 card must provide “clock” to our system.)

• CAS - Frame Type: ESF/B8ZS or D4/AMI

• PRI - Frame Type: ESF/B8ZS or D4/AMI
Switch Types: ATT 4ESS, ATT 5ESS, DMS100, DMS250, and NI2
(Sonexis ConferenceManager can be either network- or client- side.)

DialPlan- Configure the dial plan on the PBX to route calls to the Sonexis. Normally this
includes associating an extension and DID with the trunk group (be it T1, PRI or VOIP).

General Notes:
1. Customer must provide T1 straight through or crossover cable(s) to connect Sonexis
ConferenceManager to the PBX. Cable lengths in excess of 133 feet require a CSU/DSU.

2. Sonexis ConferenceManager only supports E&M Wink on CAS circuits and requires a minimum
Wink duration of 200 milliseconds.

3. Sonexis ConferenceManager DOES NOT support:
a. Shared D-channels across multiple PRIs (NFAS); or
b. QSIG (Q Signaling).

STAR Performance Kit – Pre-Implementation Site Requirements (05-28-09).Doc Page 1 of 3

VOIP systems:

General VOIP considerations:
The Sonexis transmits 20ms RTP audio via G711. We support DSCP priority tagging, expedited
forwarding.

H.323

Sonexis ConferenceManager supports the standard implementation of H.323., We support ‘fast start’
and DTMF via “user information” messages.

SIP

Sonexis ConferenceManager supports the standard implementation of SIP. We support DTMF via
“SIP INFO” and “RFC2833”. DTMF payload type is configurable 96-105

Bandwidth notation: Sonexis ConferenceManager currently supports G711 which utilizes about 90
kpbs per call in each direction, or 180 kbps total per caller.

Network Configuration:

1. IP Addresses—Sonexis ConferenceManager requires a static IP address. (A customer-specified,
friendly HOST name is strongly recommended.) Configure your Domain Name Server (DNS) so
that your network uses the same domain name whether Sonexis Conference Manager is accessed
from inside or outside your network. This is required to ensure that ConferenceManager’s
automatic email invitations will contain the correct URL for Web conferences. In addition, Sonexis
ConferenceManager will likely require a public IP address for access outside your local area
network. This public IP address can be assigned directly to the Sonexis ConferenceManager; or
you may use 1-to-1 Network Address Translation (NAT) via a firewall or router.

2. Operating System Updates—DNS should be configured so that your Sonexis ConferenceManager
can connect to Microsoft for updates using the operating system’s “Automatic Updates” utility.
Updates should be downloaded automatically and set to “notify me when downloads are ready to
install.” Since some updates require a system restart, it is strongly recommended that this
operation is performed during non-business hours.


3. Network Topology—Determine where the Sonexis ConferenceManager will reside within your
network topology (i.e., behind the firewall or in a DMZ), keeping in mind the security ramifications
of each configuration.

4. Firewall Port Configuration—Sonexis ConferenceManager requires the following firewall port
configuration:

a. Port 80: Provides Web access to the system.
b. Port 8097: Provides administrative interface access.
c. Port 9000, 9001-900x: Provides improved performance (optional). If this feature is
enabled, a number of ports (in the 9001-900x range) equal to the
anticipated number of simultaneous web conferences must be
opened. For example, ten simultaneous web conferences would
require ports 9000 and 9001-9010.

5. SSL—To use Sonexis ConferenceManager over the Secure Sockets Layer (SSL), obtain a security
certificate prior to installation. The certificate can be generated internally or obtained from a
certificate provider (e.g., VeriSign). SSL requires access on port 443. Port 80 should be closed.
(Port 8097 will still be required for access to the administrative interface.)

6. Automatic Email Invitations—Sonexis ConferenceManager can automatically send email
invitations to participants when conferences are scheduled. This functionality requires the
specification of an SMTP mail gateway. Alternatively, the SMTP server on the Sonexis
ConferenceManager can be enabled and configured to send mail directly from the server.

7. Virus Protection—A high quality antivirus program should be installed on the Sonexis
ConferenceManager. Virus definitions should be updated regularly. The following directories
should be excluded from the AV realtime scan: C:sonexis and C:Program FilesMicrosoft SQL
ServerMSSQLData

Important Note: Please read this document carefully. Contact your PBX manufacturer or support
provider to obtain all of the required components for integration with the Sonexis ConferenceManager
prior to your scheduled installation date. Sonexis representatives will not configure your PBX.

For additional information please contact Sonexis Customer Care at (866) 676-6394 or
customercare@sonexis.com.


DIGITAL CIRCUIT
T1/CAS or ISDN/PRI
Internal
Phones

PSTN

PBX (w/ available T1 cards)

“TIE LINE”
Connections
PRI or CAS
ADMIN PORT:
(Standard T1 X-
8097
over cable)
TERMINAL SERVICES
Sonexis Conference 1 Per SPAN
ENABLED
Manager

SONEXIS CONFERENCE
PSTN 2U `
MANAGER
Configuration:
“Behind a PBX” LAN Connection: External Web Client
10/100/1000
(Can be Minimal Port required:
connected via PORT 80
DMZ) (Can be configured to use
SSL)
Additional Ports:
9000
9001 thru 900x
(for Application
Sharing...optional)

`
INTERNET
INTERNAL LAN

FIREWALL EDGE ROUTER
`

Internal Web Clients

DIGITAL CIRCUIT
T1/CAS or ISDN/PRI
Internal
Phones

PSTN
PBX
T1 Endpoint (SMART
JACK)
DIGITAL CIRCUIT
T1/CAS or ISDN/PRI

CSU / DSU (if necessary)
ADMIN PORT:
8097
TERMINAL SERVICES
Sonexis Conference ENABLED
Manager

SONEXIS CONFERENCE
PSTN 2U `
MANAGER
Configuration:
“Direct TDM” LAN Connection: External Web Client
10/100/1000
(Can be Minimal Port required:
connected via PORT 80
DMZ) (Can be configured to use
SSL)
Additional Ports:
9000
9001 thru 900x
(for Application
Sharing...optional)

`
INTERNET
INTERNAL LAN

FIREWALL EDGE ROUTER
`

Internal Web Clients

Sonexis Conference
Manager

Internal
VoIP ` Phones
Configuration:
“SIP/H.323/TAPI
WAV” Internal Web Clients

INTERNAL LAN

`
LAN Connection:
10/100/1000

NOTE: Sonexis
recommends that the bridge
and the PBX be on the
ADMIN PORT: same physical switch and
8097 the the network allow for `
TERMINAL SERVICES Voice Prioritization.
ENABLED Concerning bandwidth, we
only support G.711
External Web Client
Minimal Port required:
PORT 80
SONEXIS CONFERENCE (Can be configured to use
MANAGER SSL)
Additional Ports:
9000
2U 9001 thru 900x
(for Application
Sharing...optional)

INTERNET
100/1000
IP-PBX ETHERNET
SWITCH FIREWALL EDGE ROUTER

Avaya Solution & Interoperability Test Lab

Application Notes for Configuring Sonexis
ConferenceManager with Avaya IP Office using an
ISDN/PRI trunk – Issue 1.0

Abstract

These Application Notes describe the procedure for configuring Sonexis ConferenceManager
to interoperate with Avaya IP Office using an ISDN/PRI trunk.

Sonexis ConferenceManager is an in-house audio conferencing bridge that eliminates the
costly pay-as-you-go fees of subscription-based services, while setting new standards for
security and ease of use. Sonexis ConferenceManager is designed to work within existing
voice and data networks, and Sonexis ConferenceManager is available with a fully integrated
Web conferencing option

Information in these Application Notes has been obtained through DevConnect compliance
testing and additional technical discussions. Testing was conducted via the DevConnect
Program at the Avaya Solution and Interoperability Test Lab.

CRK; Reviewed: Solution & Interoperability Test Lab Application Notes 1 of 16
SPOC 9/10/2011 ©2011 Avaya Inc. All Rights Reserved. Sonexis-IPO_PRI

1. Introduction
(herein referred to as ConferenceManager) to interoperate with Avaya IP Office.

ConferenceManager is an in-house audio conferencing bridge that eliminates the costly pay-as-
you-go fees of subscription-based services, while setting new standards for security and ease of
use. ConferenceManager is designed to work within existing voice and data networks, and
ConferenceManager is available with a fully integrated Web conferencing option.

These Application Notes assume that Avaya IP Office is already installed and basic
configuration steps have been performed. Only steps relevant to this compliance test will be
described in this document.
 PRI line configuration in IP Office
 Short Code for call route
 Incoming Call Route

2. General Test Approach and Test Results
The general test approach was to place calls to and from ConferenceManager. The main
objectives were to verify the following:
 Inbound calls
 Outbound calls
 Hold / Resume
 Call termination (origination/destination)
 Transfer (blind/consult)
 Conference (client initiated/host initiated)
 DTMF
 ANI/DNIS

2.1. Interoperability Compliance Testing
The interoperability compliance testing included features and serviceability tests. The focus of
the compliance testing was primarily on verifying the interoperability between
ConferenceManager and Avaya IP Office.

2.2. Test Results
The test objectives were verified. For serviceability testing, ConferenceManager operated
properly after recovering from failures such as cable disconnects, and resets of

2.3. Support
Technical support for the ConferenceManager solution can be obtained by contacting Sonexis:
 URL – CustomerCare@sonexis.com
 Phone – (866) 676-6394


3. Reference Configuration
Figure 1 illustrates the configuration used in these Application Notes. The sample configuration
shows an enterprise with Avaya IP Office. Endpoints include an Avaya 1616-I IP Telephone, a
4625SW IP Telephone, and an Avaya 1416 Digital Telephone on IP Office.

Note: An Avaya S8300D Server and an Avaya G450 Media Gateway were included to simulate
PSTN calls.

Figure 1: Test Configuration of Sonexis ConferenceManager


4. Equipment and Software Validated
The following equipment and software were used for the test configuration.

Equipment Software/Firmware
Avaya IP Office 500 V2 7.0(12)
Avaya IP Office Manager on Windows XP 9.0(3)
Professional 2002 with SP3
Avaya S8300D Server w/ G450 Media Gateway 6.0.1
(used to simulate PSTN calls)
Avaya H.323 IP Telephones on IP Office
4625SW (H.323) 2.9.1
1616-I (H.323) 1.22
Avaya 1416 Digital Telephone -
Avaya H.323 IP SIP Telephones on Avaya
Aura ® Communication Manager (simulating
PSTN phones)
9620 (SIP) 2.6.4
9630 (SIP) 2.6.4
9620 (H.323) 3.1
9630 (H.323) 3.1
9650 (H.323) 3.1
Sonexis on Windows Server 2008 with SP 2 10.0


5. Configure Avaya IP Office
This section describes the steps required for configuring Avaya IP Office. During the
compliance test, a PRI line was utilized between Avaya IP Office and ConferenceManager.

The procedures include the following areas:
 Verify PRI line Channels License
 Configure PRI Line
 Create the static PRI line
 Configure a short code to route calls through the PRI line
 Create an Incoming Call Route for the Inbound PRI calls

These steps are performed from the Avaya IP Office Manager.


5.1. Verify PRI Line Channels License
IP Office is configured via the IP Office Manager application. Log into the PC running the
Avaya IP Office Manager application, and select Start  All Programs  IP Office 
Manager to launch the Manager application. Select the proper IP Office system if there are
more than one IP Office system, and log in with the appropriate credentials.

From the configuration tree in the left pane, select License  IP500 Universal PRI (Additional
Channels). Verify that the License Status field is set to Valid.

If a required feature is not enabled or there is insufficient capacity, contact an authorized Avaya
sales representative to make the appropriate changes.


5.2. Configure PRI Line
From the configuration tree in the left pane, click on Line, and select 5, which is a PRI line, to
display the PRI 24 (Universal) – Line 5 screen in the right pane. Select the PRI 24 Line tab
and provide the following information:
 Switch Type – Select NI2 using the drop-down menu. During the compliance test, NI2
was utilized on both (IP Office and ConferenceManager).
 Channel Allocation – Select 231 (or 123) using the drop-down menu.
 Framing – Select ESF using the drop-down menu
 Zero Suppression – Select B8ZS using the drop-down menu
 Line Signaling - Select CPE using the drop-down menu. The ConferenceManager side
was set to Network.


Select the Channels tab to display channels. Select channels that will be used and click the Edit
button. All 23 channels were utilized during the test.


On the Multiple Channel Edit screen, provide the following information:
 Incoming Group – Enter the incoming line, created in Section 5.2.
 Outgoing Group – Enter the outgoing line, created in Section 5.2.
 Admin – Select In Service using the drop-down menu.

Click on the OK button.


5.3. Configure a Short Code to Route Calls through the PRI line
Select Short Code in the left panel. Right click and select Add. Enter 77301; where extension
77301 will be routed to ConferenceManager, in the Code text box. Select Dial for the Feature
field. Enter the Outgoing Group number created in Section 5.2 for the Line Group Id field.
Enter „.’ for the Telephone Number field. Use default values for all other fields. Click the OK
button.

Note: When extension 77031 was dialed, the call routed thru the PRI line 5.


5.4. Create an Incoming Call Route for the Inbound PRI Calls
Select Incoming Call Route in the left pane. Right-click and select New.
Enter the following:
 Any Voice for the Bearer Capability field.
 Enter the Incoming Group number created in Section 5.2 in the Line Group Id field.
 Use default values for all other fields.


Next, navigate to the Destinations tab and enter “.” under the Destination field.

Click the OK button.

After making the changes, click on the floppy disk icon (not shown) to push the changes to the
IP Office system and have them take effect

Note: Changes will not take effect until this step is completed. This may cause a reboot of
Avaya IP Office causing service disruption.


6. Configure the Sonexis ConferenceManager
Sonexis installs, configures, and customizes the ConferenceManager application for their end
customers. Thus, this section only describes the interface configuration, so that
ConferenceManager can talk to Avaya IP Office. By the request of Sonexis, the only codec
tested during the compliance test was G.711MU.
The procedures for setting up ConferenceManager for a PRI line include the following areas:
 Installing License
 Configure Telephony

6.1. Install PRI Line license
Launch a web browser, enter https://<IP address of ConferenceManager>:8097 in the URL,
and log in with the appropriate credentials. Navigate to the License menu. Enter an appropriate
license for the PRI line in the New License Key field.

Click on the Apply button.

Note: During the test, Sonexis provide the licenses for PRI, H323 and PRI lines.


6.2. Configure Telephony
Select the Telephony tab and provide the following information:
 Circuit Type – Select ISDN PRI using the drop-down menu.
 Switch Type – Select NI2 using the drop-down menu.
 Check on the Network Side box.
 Frame/Line Type – Select ESF/B8ZS using the drop-down menu.

Click on the restart telephony button.


7. Verification Steps
This section provides the tests that can be performed to verify proper configuration of Avaya IP
Office and ConferenceManager.

7.1. Verify Avaya IP Office
From a PC running the Avaya IP Office Monitor application, select Start  All Programs 
IP Office  System Status to launch the application. From the Avaya IP Office System
Status screen, select Trunks  Line 5 from the left pane and verify the trunk is Idle under the
Current State field.

8. Conclusion
These Application Notes describe the procedures required to configure Sonexis
ConferenceManager to interoperate with Avaya IP Office through a PRI trunk. Sonexis
ConferenceManager successfully passed compliance testing.

9. Additional References
The following Avaya product documentation can be found at http://support.avaya.com
[1]IP Office 7.0 Standard Version Installation, Issue 23k, May 2011, Document Number 15-
601042
[2]IP Office Release 7.0 Manager 9.0, Issue 26h, May 2011, Document Number 15-601011

Sonexis product documentation can be requested at the following site:
http://www.sonexis.com/access/index.asp?id=40&Program=DevConnect



ConferenceManager with Avaya IP Office using a SIP trunk
– Issue 1.0

Abstract

to interoperate with Avaya IP Office using a SIP trunk.

voice and data networks — and available with a fully integrated Web conferencing option


SPOC 9/20/2011 ©2011 Avaya Inc. All Rights Reserved. Sonexis-IPO_SIP

1. Introduction


 SIP trunk configuration in IP Office

 Inbound calls
 Outbound calls
 Hold / Resume
 DTMF
 ANI/DNIS


2.2. Test Results

2.3. Support
 Phone – (866) 676-6394



PSTN calls.




4625SW (H.323) 2.9.1
1616-I (H.323) 1.22
Avaya H.323 IP SIP Telephones on Avaya
Aura ® Communication Manager (simulating
PSTN phones)
9620 (SIP) 2.6.4
9630 (SIP) 2.6.4
9620 (H.323) 3.1
9630 (H.323) 3.1
9650 (H.323) 3.1


compliance test, a SIP trunk was utilized between Avaya IP Office and ConferenceManager.

 Verify SIP trunk Channels License
 Configure LAN interface
 Enable SIP Trunk
 Create the static SIP line
 Configure SIP URI parameters for the SIP Line
 Configure VoIP Parameters for the SIP Line
 Configure a short code to route calls through the SIP trunk
 Create an Incoming Call Route for the Inbound SIP calls



5.1. Verify SIP Trunk Channels License
IP Office is configured via the IP Office Manager application. Log into the PC running the
Avaya IP Office Manager application, and select Start  All Programs  IP Office 
Manager to launch the Manager application. Select the proper IP Office system if there are
more than one IP Office system, and log in with the appropriate credentials.

From the configuration tree in the left pane, select License  SIP Trunk Channels to display
the SIP Trunk Channels screen in the right pane. Verify that the License Status field is set to
Valid.

If a required feature is not enabled or there is insufficient capacity, contact an authorized Avaya
sales representative to make the appropriate changes.


5.2. Configure LAN interface
From the configuration tree in the left pane, select System to display the System screen in the
right pane. Click the LAN1 tab. Under the LAN1 tab, select the LAN Settings sub-tab, and
provide IP Address and IP Mask.


5.3. Enable SIP Trunk
Under the LAN1 tab, select the VoIP sub-tab and check the SIP Trunks Enable box. Click the
OK button.

Note: During the initial configuration of Avaya IP Office, the LAN1 was configured as a private
network (LAN) and the LAN2 was configured as a public network (WAN). Avaya IP Office can
support SIP extensions on the LAN1 and/or LAN2 interfaces. However, the compliance test
used the LAN1 interface for a SIP trunk termination.


5.4. Create SIP Lines for a SIP Trunk
Select Line in the left pane. Using the right mouse click, select New  SIP Line [not shown],
and create a new Line Number. Enter an appropriate domain on the ITSP Domain Name field,
and check the InService check box. During the compliance test, a SIP line (27) was configured.


Select the Transport sub-tab, and provide the following information:
 ITSP Proxy Address – Enter the IP address of the far-end SIP termination point.
 Layer 4 Protocol – Select UDP.
 Use Network Topology Info – Select None.


5.5. Configure SIP URI Parameters for the SIP Line
Select the SIP URI tab to configure SIP URI parameters for the SIP Line. Click on the Add
button.

Select * for the Local URI, Contact, and Display Name fields. Enter a unique number for the
Incoming Group and Outgoing Group fields. The Incoming Group field will be used for
mapping inbound calls from the SIP trunk to local stations. The Outgoing Group will be used
for routing calls externally via the Short Code configured in Section 5.7. Use default values for
all other fields. Click the OK button.


5.6. Configure VoIP Parameters for the SIP Line
Select the VoIP tab to Configure VoIP parameters for the SIP Line. For Compression Mode,
select G.711 ULAW 64K, since ConferenceManager only supports G.711MU. Check Re-invite
Supported check boxes.



5.7. Configure a Short Code to Route Calls through the SIP trunk
Select Short Code in the left panel. Right click and select Add. Enter 77301; where extension
77301 will be routed to ConferneceManager, in the Code text box. Select Dial for the Feature
field. Enter the Outgoing Group number created in Section 5.5 for the Line Group Id field.
Enter .”@avaya.com” for the Telephone Number field. Use default values for all other fields.

Note: When extension 77031 was dialed, the call routed thru the SIP trunk 27.


5.8. Create an Incoming Call Route for the Inbound SIP Calls
 Enter the Incoming Group number created for the URI in Section 5.5 in the Line Group Id
field.


Next, navigate to the Destinations tab and enter ―.‖ under the Destination field.





The procedures for setting up ConferenceManager for a SIP trunk include the following areas:

6.1. Install SIP Trunk license
license for SIP trunk in the New License Key field.


Note: During the test, Sonexis provide the licenses for SIP, H323 and PRI trunks.


 Dialout Gateway IP: Enter the far-end SIP trunk termination point. During the
compliance test, it should be the IP Office LAN1 IP address.
 DTMF Payload Type: Set the payload type to 101





8. Conclusion
ConferenceManager to interoperate with Avaya IP Office through a SIP trunk. Sonexis

601042




ConferenceManager with Avaya IP Office using an H323
trunk – Issue 1.0

Abstract

to interoperate with Avaya IP Office using a H323 trunk.

voice and data networks — and available with a fully integrated Web conferencing option


SPOC 9/7/2011 ©2011 Avaya Inc. All Rights Reserved. Sonexis-IPOH323

1. Introduction


 H323 trunk configuration in IP Office

 Inbound calls
 Outbound calls
 Hold / Resume
 DTMF
 ANI/DNIS


2.2. Test Results

2.3. Support
 Phone – (866) 676-6394



PSTN calls.




4625SW (H.323) 2.9.1
1616-I (H.323) 1.22
Avaya H.323 IP Telephones on Avaya Aura ®
Communication Manager (simulating PSTN
phones)
9620 (SIP) 2.6.4
9630 (SIP) 2.6.4
9620 (H.323) 3.1
9630 (H.323) 3.1
9650 (H.323) 3.1


compliance test, a H323 trunk was utilized between Avaya IP Office and ConferenceManager.

 Configure LAN interface
 Enable H323 Gatekeeper
 Create the static H323 line
 Configure a short code to route calls through the H323 trunk
 Create an Incoming Call Route for the Inbound calls



5.1. Configure LAN interface
From the configuration tree in the left pane, select System to display the System screen in the
right pane. Click the LAN1 tab. Under the LAN1 tab, select the LAN Settings sub-tab, and
provide IP Address and IP Mask.


5.2. Enable H323 Gatekeeper
Under the LAN1 tab, select the VoIP sub-tab and verify the H323 Gatekeeper Enable box.

Note: During the initial configuration of Avaya IP Office, the LAN1 was configured as a private
network (LAN) and the LAN2 was configured as a public network (WAN). Avaya IP Office can
support H323 extensions on the LAN1 and/or LAN2 interfaces. However, the compliance test
used the LAN1 interface for a H323 trunk termination.


5.3. Create a Line for a H323 Trunk
Select Line in the left pane. Using the right mouse click, select New  H323 Line [not shown],
and create a new Line Number, using the drop-down menu. Enter an appropriate Incoming
Group ID, Outgoing Group ID, Number of Channels, Outgoing Channels, and Voice
Channels.

During the compliance test, a H323 line (25) was configured.


Select the VoIP Settings sub-tab, and provide the following information:
 Gateway IP Address – Enter the IP address of the far-end H323 termination point.
 Compression Mode – Select G.711 ULAW 64K, using the drop-down menu.
 Supplementary Services– Select H450, using the drop-down menu. To work with
ConferenceManager, IP Office should select H450 as a Supplementary Services to
enable the fast start option.
 Check the following check boxes
o Enable Fast Start
o Out of Band DTMF
 Make sure the Allow Direct Media Path field is not checked. For the H323 trunk
solution, ConferenceManager only supports Shuffling Off (no direct media path).

Click on the OK button


5.4. Configure a Short Code to Route Calls through the H323 trunk
Select Short Code in the left panel. Right click and select Add. Enter 77301 in the Code field;
where extension 77301 will be routed to ConferneceManager. Enter . for the Telephone
Number field. Enter the Outgoing Group number created in Section 5.3 for the Line Group
Id field. Use default values for all other fields.


Note: When extension 77031 was dialed, the call routed thru the H323 trunk 25.


5.5. Create an Incoming Call Route for the Inbound H323 Calls
 Enter the Incoming Group number created for the URI in Section 5.3 in the Incoming
Group Id field.


Next, navigate to the Destinations tab and enter ―.‖ under the Destination field.





The procedures for setting up ConferenceManager for a H323 trunk include the following areas:

6.1. Install H323 Trunk license
license for H323 trunk in the New License Key field.


Note: During the test, Sonexis provide the licenses for SIP, H323 and PRI trunks.


 Dialout Gateway IP: Enter the far-end H323 trunk termination point. During the
compliance test, it should be the IP Office LAN1 IP address.
 H.323 Type: Set the type to Definity, using the drop-down menu.





8. Conclusion
ConferenceManager to interoperate with Avaya IP Office through a H323 trunk. Sonexis

601042



©2011 Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and ™
are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the
property of their respective owners. The information provided in these Application Notes is
subject to change without notice. The configurations, technical data, and recommendations
provided in these Application Notes are believed to be accurate and dependable, but are
presented without express or implied warranty. Users are responsible for their application of any
products specified in these Application Notes.

Please e-mail any questions or comments pertaining to these Application Notes along with the
full title name and filename, located in the lower right corner, directly to the Avaya DevConnect
Program at devconnect@avaya.com.


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