SIP Connect for ShoreTel UC with AudioCodes ESBC Customer

Transcription

IN-15068
AUTHOR’S NAME: SHORETEL
BUSINESS UNIT: SHORETEL AUSTRALIA
SUB BUSINESS UNIT:
ISSUE DATE: 3TH
NOV 2015 | ISSUE NUMBER: 1.0
DOCUMENT NO:
AQG-1963
SIP CONNECT FOR SHORETEL UNIFIED COMMUNICATIONS WITH
AUDIOCODES ESBC
CUSTOMER CONFIGURATION GUIDE
SUMMARY: This document describes the configuration required to setup the ShoreTel Unified
Communication Platform and AudioCodes Mediant 800/1000 SBC solution. The topology
involves deploying the AudioCodes Session Border Controller to allow connection of local
ShoreTel IP users to Telstra SIP Connect.
VERSION LABEL: FINAL
SECURITY CLASSIFICATION: Telstra Unrestricted
TELSTRA CORPORATION LIMITED (ABN 33 051 775 556) |
FINAL| | SHORETEL_UC_AUDIOCODES_ESBC CONFIGURATIONGUIDE
TABLE OF CONTENTS
1.
Introduction ShoreTel UC with Audiocodes ESBC ..............................................................4
1.1. PURPOSE ............................................................................................................4
1.2. AUDIENCE ..........................................................................................................4
1.3. SOLUTION OVERVIEW ..........................................................................................5
2.
1.3.1.
BENEFITS OF SOLUTION ...........................................................................5
1.3.2.
LIMITATIONS ..........................................................................................6
1.3.3.
TELSTRA ACCREDITED TEST CONFIGURATION ..............................................7
Telstra Supplied Information ........................................................................................8
2.1. TELSTRA PRODUCTION INFORMATION ....................................................................8
2.2. NATIONAL SBC ADDRESSES ..................................................................................8
2.3. NATIONAL DNS ADDRESSES ..................................................................................9
2.4. PING-ABLE NETWORK ADDRESSES .........................................................................9
3.
General Configuration ................................................................................................ 10
3.1. OVERVIEW ........................................................................................................ 10
4.
ShoreTel Configuration .............................................................................................. 11
4.1. INTRODUCTION ................................................................................................. 11
4.2. SHORETEL CONFIGURATION STEPS ...................................................................... 11
4.2.1.
CALL CONTROL SETTINGS ....................................................................... 11
4.2.2.
SITES SETTINGS .................................................................................... 13
4.2.3.
SWITCH SETTINGS – ALLOCATING PORTS ................................................. 14
4.2.4.
SYSTEM SETTINGS – TRUNK GROUPS ....................................................... 16
4.2.5.
SYSTEM SETTINGS – INDIVIDUAL TRUNKS .................................................... 19
4.2.6.
SYSTEM SETTINGS – CUSTOM DIAL PLAN .................................................... 21
4.3. AUDIOCODES CONFIGURATION STEPS .................................................................. 25
4.3.1.
GENERAL INFORMATION ......................................................................... 25
4.3.2.
DEFINE AN IP ADDRESS FOR THE INTERNAL AND EXTERNAL INTERFACE ........ 25
4.3.3.
ENABLE THE SBC APPLICATION ................................................................ 27
4.3.4.
DEFINE NUMBER OF MEDIA CHANNELS ...................................................... 28
4.3.5.
RESET AUDIOCODES ESBC ...................................................................... 28
4.3.6.
CONFIGURE SIP AND RTP INTERFACES ...................................................... 29
4.3.7.
DEFINE PROXY SETS ............................................................................... 32
4.3.8.
CREATE IP PROFILES .............................................................................. 33
4.3.9.
DEFINE IP GROUPS ................................................................................. 35
4.3.10. CONFIGURE ACCOUNT TABLE FOR TELSTRA SIP CONNECT REGISTRATION ..... 36
4.3.11. DEFINE IP-TO-IP ROUTING RULES ............................................................ 36
FINAL | | SHORETEL_UC_AUDIOCODES_ESBC CONFIGURATIONGUIDE
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4.3.12. DEFINE MANIPULATION RULES ................................................................. 38
4.3.13. E-SBC SECURITY .................................................................................... 41
5.
Vendor Support ........................................................................................................ 42
6.
REFERENCES ........................................................................................................... 42
7.
DEFINITIONS ........................................................................................................... 42
8.
ATTACHMENTS ........................................................................................................ 42
9.
Apendix A – Audiocodes M1000 MSBG Interface Setup ................................................... 43
9.1.1.
DEFINE IP ADDRESSES FOR OAMP, MEDIA AND CONTROL ............................ 43
10. DOCUMENT CONTROL SHEET ..................................................................................... 45
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1.
INTRODUCTION SHORETEL UC WITH AUDIOCODES ESBC
1.1. PURPOSE
This document is intended to give an overview of the configuration tasks required in order for
Telstra IP Telephony (TIPT) and ShoreTel UC with Audiocodes ESBC to form an integrated solution.
It is not the intent of this document to detail all the configuration steps required to configure the
PBX for full operation at a customer site, only the additional steps/settings required to integrate it
into the TIPT environment. For example, application server and session Border controller settings,
etc
Configuration is provided for ShoreTel 14.2 & AudioCodes ESBC v6.6 (Median 800/1000) for SIP
Trunking to the Telstra IP Telephony (TIPT) platform.
Note: Detailed configuration documentation and specific information that relates to this IP PBX is
only available from the vendor or its licensed partners and re-sellers. It is not supplied as
part of the Vendor Self Certification program.
1.2. AUDIENCE
The intended audience for this document is:
•
•
•
Telstra & Customer Technical Administrators,
Vendor representatives, and
Technical support staff
Who may be responsible for appropriate configuration and assurance of ShoreTel solutions for use
with Telstra SIP Connect (including Network & IT)
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1.3. SOLUTION OVERVIEW
Telstra IP Telephony’s SIP Connect for ShoreTel's Unified Communications with Audiocodes ESBC
provides the facility for the delivery of SIP trunks via the Telstra network without the requirement
for ISDN services at the customer site. Calls both to and from the PSTN can be made across the
configured SIP Trunk subject to ShoreTel & AudioCodes configuration.
Figure 1: Basic SIP Connect for TIPT Architecture
1.3.1. BENEFITS OF SOLUTION
SIP Trunking allows the use of Session Initiation Protocol (SIP) communications instead of the
typical analog, Basic Rate Interface (BRI) or E1 trunk connections. Having the pure IP trunk to the
Service provider allows for more control and options over the communication link. This application
note provides the details on connecting the ShoreTel® IP phone system through an Audiocodes
ESBC Gateway which is connected to both the LAN and WAN and acts as a secure gateway to
Telstra for SIP Trunking.
ShoreTel and Audiocodes have teamed up to build a solid security focused solution, ShoreTel being
the IP PBX which sits on the LAN and connects to the AudioCodes Gateway, providing a solution to
allow customers the ability to connect to SIP Trunks offered by Telstra Australia in a secure
manner. The Audiocodes ESBC Gateway then is connected to not only the LAN but also the WAN,
providing the typical firewall security abilities but also intelligent SIP routing and such SIP features
as:
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•
•
•
•
•
•
Registration
Digest Authentication
Dial Plan Modification
Back to Back User Agent (Terminates SIP messaging on both LAN and WAN side for SIP
Protocol Normalization)
Transfer conversion of SIP REFER to SIP reINVITE messaging
Quick configuration templates for each of the certified ITSPs
1.3.2. LIMITATIONS
The limitations of the integrated SIP Connect solution between TIPT and ShoreTel/AudioCodes are:
•
SIP Trunk Call Capacity is controlled by both SIP Connect (Trunk Group Call Capacity) and
ShoreTel/AudioCodes. Careful consideration must be undertaken to ensure that the SIP
Connect implementation is designed correctly.
Off-net calls forwarded or transferred off-net hairpin via AudioCodes ESBC, this results in
two trunks calls being consumed as well as extra bandwidth.
• ShoreTel vTrunk Switches are not currently supported/validated with Telstra
SIP Connect. It will be supported in future releases.
Please consult the ShoreTel Administration Guide (V14.2), Section 18 – Session Initiation Protocol.
Here is an excerpt of section 18.2.5.
•
General SIP Feature Considerations
This section describes various features or functions that ShoreTel SIP supports in the current
release as well as functions that ShoreTel SIP does not support.
• In the current release, the following features are supported by SIP only if the trunk has a
SIP trunk profile with hairpinning and the trunk is on a halfwidth switch:
o Silent Coach
o Silent Monitor
o Barge-In
o Call recording 3-way conference on a SIP trunk call uses Make Me conference ports.
A minimum of 3 Make Me ports must be configured to support 3-way conferencing.
Make Me conferencing for 4 to 6 parties is not supported.
• Fax (and modem) redirection works only if the carrier or ITSP supports T.38. For details
about T.38, see "T.38 Support on ShoreTel Switches" on page 134 of the ShoreTel
Administration Guide. Telstra does not support T.38, therefore fax redirection is not
supported with Telstra SIP Connect trunks.
• The maximum number of jack based music on hold (MOH) streams that a SIP-enabled
switch can support varies with the switch model and the switch’s configuration. Also, the
allotment of resources for jack-based MOH includes streams for Backup Auto Attendant and
transmission of ringback tones. The range of such streams across all the voice switch
models is 14–60.
NOTE For SIP trunks to transport jack-based MOH, the MOH source is the SIP
switch, as follows: An external source for MOH plugs into the SIP switch at the
switch’s MOH jack, and the switch places the stream on the trunk.
• A SIP switch attempts to transmit MOH over G.711 U. (Switches supports G.711 A-law and
U-law.) If the far end does not support G.711, the switch uses G.729.
• If Make Me conferences are planned, a minimum of four Make Me ports must be reserved.
A three-way Make Me conference uses three Make Me ports, a four-way conference uses
four ports, and so on up to the maximum of a six-way conference. For each media stream,
up to the maximum of six-way conferencing, an additional Make Me conference port must
be available.
• End-users can set up Make Me conference calls by using their ShoreTel Communicator. Like
extensions with support of Media Gateway Control Protocol (MGCP), SIP extensions require
permissions and a minimum of four Make Me ports to set up Make Me conference call.
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•
A SIP trunk can be a member of a three-party conference but cannot initiate a three-way
conference (unless the SIP device merges the media streams).
1.3.3. TELSTRA ACCREDITED TEST CONFIGURATION
This Configuration Guide is based on
ShoreTel release 14.2
AudioCodes ESBC release 6.6
Broadsoft Version R17 SP2
Telstra document “TAF001-711434 Test Environment specifications”.
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2.
TELSTRA SUPPLIED INFORMATION
There are a number of parameters Telstra will supply in the RASS order which will need to be
entered into ShoreTel Director and the AudioCodes Web Admin interface.
The Network Server Redirection solution has now been deployed in the TIPT Production Network.
When registering with the Production Network, the unit will Register with credentials Pilot
Number@customer SIP Realm/Domain in lieu of the Pilot Number@Proxy Server Domain Name.
The customer SIP Realm/Domain is then used as their Address in all SIP message (includes
Subscribes, Options, Message etc.)
2.1. TELSTRA PRODUCTION INFORMATION
A RASS (Records Automation for Special Services) order will be generated by Telstra for each new
customer which details all the specific parameters for that customer.
RASS Order Example
Enterprise: N3301000R
Hosted Group: N9876543R
Trunk Group: N3301011R
Authentication User ID: N3301011R
Authentication password: Password1!
Trunk Pilot Number: 0353950600
SIP Realm/Domain: sipconn.test1.com
DME: AIU-9999
CPE Ordered:
There is no CPE to be ordered
Number range: 0353950600-604, 0353950920-24
Number of SIP Channels:
Optional Packs: YES
Business Continuity destination number: N5544332R
Note: When entering the Pilot Number for Registration in the Production Network,
ONLY 9 digits are required. The leading “0” is dropped. (i.e. 0396241990 use
396241990 to register)
2.2. NATIONAL SBC ADDRESSES
SBC s’ are positioned between the customer VPN and the TIPT core. They provide firewalling
features and a logical separation between the Networks. The SBC is treated as the SIP Server
from the perspective of the PBX Gateway instead of the actual BroadWorks SIP Server. There are
SBC’s located in all states and they are in fully redundant deployment. The SBC address MUST
be entered as an FQDN not an IP address.
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Victoria
sbc-vic.nipt.telstra.com
NSW
sbc-nsw.nipt.telstra.com
Qld
sbc-qld.nipt.telstra.com
NT
sbc-nt.nipt.telstra.com
WA
sbc-wa.nipt.telstra.com
SA
sbc-sa.nipt.telstra.com
Tas
sbc-tas.nipt.telstra.com
Act
sbc-act.nipt.telstra.com
2.3. NATIONAL DNS ADDRESSES
DNS Addresses are:
Primary
203.52.0.221
Secondary
203.52.1.222
2.4. PING-ABLE NETWORK ADDRESSES
These addresses can be used to verify connectivity when configuring the PBX.
Vic
203.52.0.161
Nsw
203.52.1.161
WA
203.52.2.161
QLD
203.52.3.161
SA
203.44.43.161
ACT
203.44.44.161
NT
203.44.42.1
TAS
203.44.42.225
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3.
GENERAL CONFIGURATION
3.1. OVERVIEW
The following points are to be observed when provisioning ShoreTel/Audiocodes to interoperate
with TIPT.
The configuration information below shows examples for configuring the ShoreTel, Audiocodes and
Telstra SIP Connect Trunks. Even though configuration requirements can vary from setup to
setup, the information provided in these steps, along with the ShoreTel Planning and Installation
Guide (or the Administration Guide) and documentation provided by Audiocodes and Telstra should
prove to be sufficient. However every design can vary and some may require more planning than
others.
The following hardware while not necessarily used during the certification itself is guaranteed to
comply with the testing results of this application note.
ShoreTel Voice switches
ShoreGear 30
ShoreGear 50
ShoreGear 50v
ShoreGear 90
ShoreGear 90v
ShoreGear 90BRI
ShoreGear 90BRIv
ShoreGear 120
ShoreGear 220E1
ShoreGear E1k
ShoreTel Phones
All officially ShoreTel phones are also compliant with the test results and supported
Analog Phones
Any analog phones connected to the following ShoreGear voice switches will be compliant
with the test results documented in this application note.
AudioCodes
When making changes for each screen described in the following steps, ensure that you
click the Submit button to save your changes; unless otherwise instructed. Some of the
changes may require that you reset the AudioCodes Gateway for these changes to take
effect. Therefore, reset the AudioCodes Gateway when necessary.
The configuration flow will be based on the worksheet as outlined on the table below. It also
applies to the screenshot enclosed on this document to provide visual reference.
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4.
SHORETEL CONFIGURATION
4.1. INTRODUCTION
This section describes the ShoreTel system configuration to support SIP Trunking. The section is divided into
general system settings and trunk configurations (both group and individual) needed to support SIP Trunking.
Note: ShoreTel basically just points its Individual SIP Trunks to the Audiocodes ESBC.
The first settings to address within the ShoreTel system are the general system settings. These configurations
include the Call Control, the Site and the Switch settings. If these items have already been configured on the
system, skip this section and go on to the “ShoreTel System Settings – Trunk Groups” section below.
4.2. SHORETEL CONFIGURATION STEPS
4.2.1. CALL CONTROL SETTINGS
The first settings to configure within ShoreW are Director are the Call Control Options. To configure these
settings for the ShoreTel system, log into ShoreW are Director and select “Administration” then “Call Control”
followed by “Options”.
1.
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Open the 'ShoreW are Director Admin' page click on (Administration > Call Control > Options):
2.
The “Call Control Options” screen will then appear.
3.
W ithin the “Call Control Options” screen, confirm that the appropriate settings are made for the
“DTMF Payload Type”, “Realm”, “Enable SIP Session Timer” and “Always Use Port 5004 for RTP”
fields.
4.
The “DTMF Payload Type (96-127)” parameter defaults to a value of 102, you will need to ensure that
you change this parameter to a value of 101 to interoperate with Telstra. Once you modify this
parameter you will need to reboot all of the ShoreTel IP Phones, not rebooting the ShoreTel IP
phones will cause the default value (102) to be utilized.
The “Realm” parameter is used in authenticating all SIP devices. It is typically a description of the
computer or system being accessed. Changing this value will require a reboot of all ShoreGear
switches serving SIP extensions. It is not necessary to modify this parameter to get the ShoreTel IP
PBX system functional with Telstra SIP Connect. Verify that the “Enable SIP Session Timer” box is
checked (enabled).
Next the Session Interval Timer needs to be set. The recommended setting for “Session Interval” is
3600 seconds. The last item to select is the appropriate refresher (from the pull down menu) for the
SIP Session Timer. The “Refresher” field will be set either to “Caller (UAC)” [User Agent Client] or to
“Callee (UAS)” [User Agent Server]. If the “Refresher” field is set to “Caller (UAC)”, the Caller’s
device will be in control of the session timer refresh. If “Refresher” is set to “Caller (UAS)”, the device
of the person called will control the session timer refresh.
The last item is to verify that the “Always Use Port 5004 for RTP” is not enabled, if this is a new
installation the option will be grayed out, do not modify this parameter if it is disabled.
5.
6.
7.
Note: Disabling (un-checking) the parameter “Always Use Port 5004 for RTP” is required for
implementing SIP on the ShoreTel system. For SIP configurations, Dynamic User Datagram Protocol
(UDP) must be used for RTP Traffic. If the box is unchecked, MGCP will no longer use UDP port 5004;
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MGCP and SIP traffic will use dynamic UDP ports. Once this parameter is unchecked, make sure that
“everything” (IP Phones, ShoreGear Switches, ShoreW are Director, Distributed Voice Services /
Remote Servers, Conference Bridges and Contact Centers) are “fully” rebooted –this is a “one time
only” item. By not performing a full system reboot, one way audio will probably occur during initial
testing.
4.2.2. SITES SETTINGS
The next settings to address are the administration of sites. These settings are modified under the
ShoreW are Director by selecting “Administration”, then “Sites”.
1.
Open the 'ShoreW are Director Admin' page click on (Administration > Sites):
2.
This selection brings up the “Sites” screen. W ithin the “Sites” screen, select the name of the site to
configure.
The “Edit Site” screen will then appear.
3.
Note: Bandwidth of 2046 is just an example. Please refer to the ShoreTel Planning and Installation
Guide for additional information on setting Admission Control Bandwidth.
Sites Edit screen – Admission Control Bandwidth
The Admission Control Bandwidth defines the bandwidth available to and from the site. This is
important as SIP trunk calls may be counted against the site bandwidth. Bandwidth needs to be set
appropriately based on site setup and configuration with Telstra SIP Connect. Please refer to the
ShoreTel Planning and Installation Guide for additional information.
Sites Edit screen – Intra / Inter-Site Calls
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By default ShoreTel has 12 built-in codecs; these codecs can be grouped as “Codec Lists” and defined
in the sites page for “Inter-site” and “Intra-site” calls. By default "Very Low Bandwidth Codecs"
contains two codecs, G.729 and G.711u, with G.729 being the primary codec of choice. Make a copy
of the “Very Low Bandwidth” codec list, define a name and add the G.711a (PCMA) codec, move it
above the PCMU entry, then save the change. It should look as follows:
Then configure the "Inter-Site Calls" option for the “Codec List” you just created and save the change.
By default the “Very High Bandwidth Codecs” list contains 8 different codecs. Make a copy of the
“Very High Bandwidth Codecs” codec list, define a name and add the G.711a (PCMA) codec, once
again, move it above the PCMU entry, then save the change. It should look as follows:
Then configure the "Intra-Site Calls" option for the “Codec List” you just created and save the change.
Note: If you wish to use the G.729 codec as your primary choice, select the G729/8000 codec and
move it up above the PCMA/8000 entry.
4.2.3. SWITCH SETTINGS – ALLOCATING PORTS
The final general settings to input are the ShoreGear switch settings. These changes are modified by
selecting “Administration”, then “Switches” in ShoreW are Director.
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1.
Open the 'ShoreW are Director Admin' page click on (Administration > Platform Hardware >
Voice Switches > Appliances > Primary): This action brings up the “Switches” screen
2.
From the “Primary Voice Switches / Service Appliances” screen simply select the name of the
switch to configure. The “Edit ShoreGear …Switch” screen will be displayed. W ithin the “Edit
ShoreGear …Switch” screen, select the desired number of SIP Trunks from the ports available.
Each port designated as a SIP Trunk enables the support for 5 individual trunks.
3.
If you will be using features which require “Media Proxy Ports” (these features include Silent Monitor
& Call Record, among others – refer to the ShoreTel Planning and Installation Guide or ShoreTel
Administration Guide for more detail about which features require Media Proxy Ports) you will need
to enable this on your SIP Trunk switch.
4.2.4. SYSTEM SETTINGS – TRUNK GROUPS
If the SIP Trunk Groups have already been configured on the system, skip down to the “ShoreTel System
Settings - Individual Trunks” section. The settings for Trunk Groups are changed by selecting “Administration”,
then “Trunks” followed by “Trunk Groups” within ShoreWare Director.
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1.
Open the 'ShoreW are Director Admin' page click on (Administration > Trunks > Trunk Groups):
This action brings up the “Trunk Group” screen
2.
From the pull down menus on the “Trunk Groups” screen, select the site desired and select the
“SIP” trunk type to configure and click on the “Go” link from “Add new trunk group at site:” The
“Edit SIP Trunk Group” screen will appear.
Trunk Group and Inbound
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1.
W ithin the “Edit SIP Trunks Group” screen define a name for the trunk group, in this example we
chose “Telstra SIP Connect”.
2.
The “Profile:” parameter is drop down selection, no modification is necessary, leave at a default
setting of “Default ITSP”
3.
The “Enable Digest Authentication” field is not required when connecting to an Audiocodes device.
4.
The “Enable SIP Info for G.711 DTMF Signaling” box should not be checked. Enabling SIP info is
currently only used with tie trunks between ShoreTel systems.
5.
The next item to change in the “Edit SIP Trunks Group” screen is to make the appropriate settings
for the “Inbound:” fields.
6.
W ithin the “Inbound:” settings ensure the “Number of Digits from CO” is set to 8 for Telstra and
ensure the “DNIS” or “DID” box is checked.
7.
The last item to define is the “Destination” parameter, this will determine where an inbound call is
routed if there isn’t a DNIS, DID or extension match, we chose the default Auto Attendant menu.
Outbound and Trunk Services
1.
If this trunk group is to be used for outbound calls, be sure to enable (check) the “Outbound:”
parameter, then define an “Access Code:” and “Local Area Code:” Please refer to the ShoreTel
Planning and Installation Guide for additional information. Enter the SIP Trunk Pilot number here.
2.
The parameter “Send Original Caller Information” should be enabled. This enables P-AssertedIdentity and Diversion headers on SIP messages from ShoreTel.
3.
The parameter “Caller ID not blocked by default” determines if the call is sent out as <unknown> or
with caller information (Caller ID). User DID etc. will impact how information is passed out to the SIP
Trunk group; this parameter needs to be enabled (checked).
4.
Set the Trunk Digit Manipulation parameters:
a. The parameter “Dial Local Numbers in National Form” needs to be disabled to properly send
calls to the Telstra network.
b. The parameter “Dial in E.164 Format” needs to be enabled to properly send calls to the Telstra
network.
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5.
After these settings are made to the “Edit SIP Trunk Group” screen, press the “Save” button to save
the changes. This completes the SIP Trunk Group configuration.
4.2.5. SYSTEM SETTINGS – INDIVIDUAL TRUNKS
This section covers the configuration of the individual trunks. Select “Administration”, then “Trunks” followed by
“Individual Trunks” to configure the individual trunks.
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1.
Open the 'ShoreW are Director Admin' page click on (Administration > Trunks > Individual
Trunks): This action brings up the “Individual Trunks” screen
2.
The “Trunks by Group” screen that is used to change the individual trunks settings then appears.
3.
Select the site for the new individual trunk(s) to be added and select the appropriate trunk group
from the pull down menu in the “Add new trunk at site” area. In this example, the site is
“Headquarters” and the trunk group is “Telstra SIP Connect”, which was created above. Click on the
“Go” link to bring up the “Edit Trunk” screen.
4.
From the individual trunks “Edit Trunk” screen, input a name for the individual trunks, in this example
below “Telstra SIP Connect” which was used.
5.
Select the appropriate switch, select the SIP Trunk type and input the number of trunks. W hen
selecting a name, the “best practice” is to name the individual trunks the same as the name of the
trunk group so that the trunk type can easily be tracked. Select the switch upon which the individual
trunk will be created.
6.
In the “IP Address:” parameter input the IP address of the Audiocodes LAN interface (usually eth0).
The last step is to select the number of individual trunks desired (each one supports “one” audio
path – example if 30 is define, then 30 audio paths can be up at one time). Once these changes are
complete, press the “Save” button to input the changes.
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Note: Individual SIP Trunks cannot span networks. SIP Trunks can only terminate on the switch
selected. There is no failover to another switch. For redundancy, two trunk groups will be needed
with each pointing to another Audiocodes device – just the same as if PRI were being used.
After setting up the trunk groups and individual trunks, refer to the ShoreTel Product Installation
Guide to make the appropriate changes for the User Group settings. This completes the settings for
the ShoreTel system side
4.2.6. SYSTEM SETTINGS – CUSTOM DIAL PLAN
This section covers the configuration of the custom dial plan to support Telstra FAC *57 (MCT Feature). Select
“Administration”, then “Trunks” followed by “Individual Trunks” to configure the individual trunks. Note: this
should only be accessed when making detailed changes to the system and please do not modify
any area that you are not familiar with.
1. Log into ShoreW are Director using the Support Entry Mode, please refer to the Knowledgebase
KB12996 for more info on accessing Support Entry Mode.
2.
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Open the 'ShoreW are Director Admin' page click on (Administration > Trunks > Trunk Groups):
This action brings up the “Individual Trunks” screen
3.
From the Trunk Groups screen, select the desired trunk to be configured. The “Edit SIP Trunk
Group” screen will appear.
4.
5.
The “Edit SIP Trunk Group” screen will appear.
W ithin the “Edit SIP Trunks Group” screen click on the “Edit Custom Dialing rule” The “Trunk Groups
Dialing Rules” screen will appear.
6.
The next step is to add the following Custom dial string to the trunk plan: <*57>X.%140G
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7.
Click Save
8.
Open the 'ShoreW are Director Admin' page click on (Administration > Sites):
9.
This selection brings up the “Sites” screen. W ithin the “Edit Site” screen click on the “Edit Site
Dialing rule” The “Site Dialing Rules” screen will appear.
10. The next step is to add the following Custom dial string to the Site plan <*57>X.%1G
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11. Click Save
4.3. AUDIOCODES CONFIGURATION STEPS
4.3.1. GENERAL INFORMATION
When making changes for each screen described in the following steps, ensure that you
click the Submit button to save your changes; unless otherwise instructed.
Some of the changes may require that you reset the AudioCodes Gateway for these
changes to take effect.
The configuration flow will be based on the worksheet as outlined on the table below. It
also applies to the screenshot enclosed on this document to provide visual reference.
4.3.2. DEFINE AN IP ADDRESS FOR THE INTERNAL AND EXTERNAL INTERFACE
The procedure below describes how to configure the IP interface for each network.
 To configure the IP interface:
PAGE 25 OF 45
1.
Open the Multiple Interface table (Configuration tab > VoIP menu > Network submenu > IP
Settings). Assign the Application Type, IP address, Prefix Length/Subnet Mask, Gateway,
VLAN ID, Interface Name and DNS Server. Take note that for the Interface Name the word
“W AN” should NOT be used as it is a reserved word. The Interface Name can be chosen
arbitrarily.
•
Add the IP interface entry for the Internal network signaling leg:
a. Application Type = OAMP + Media + Control
b. IP address = 10.8.70.1
c. Prefix Length = 16
d. Gateway = ‘only if required’
e. VLAN ID = 1
f.
g.
h.
i.
•
Interface Name = Voice
Primary DNS Server IP Address = ‘ only if required ‘
Underlying interface: Group 1
Click Apply, and then Done to validate the entry.
Add the IP interface entry for the network signaling leg:
a. Application Type = Media + Control
b.
IP address: 10.2.1.212
c.
d.
Netmask : 24
Default gateway: 10.2.1.1
e.
VLAN ID = 10
f.
Interface Name = External
g.
Primary DNS Server IP Address = 10.86.113.20
h.
i.
Underlying Interface : Group 2
Click Apply, and then Done to validate the entry.
Figure 4.3-1: Defining the IP Address
 To confirm physical port allocation :
Each pair of ports on the AudioCodes ESBC by default is allocated to different Group. To
confirm what ports are in Group 1 and Group 2 as defined in the network Interface
setting:
1.
Open connection page (Configuration tab >>VoIP menu >Network submenu >>Physical
port setting to confirm what ports are in Group 1 and Group 2 as defined in the network
Interface setting :
PAGE 26 OF 45
Figure 4.3-2: Physical port setting
NOTE:
In case different VLAN ID has to be assigned to each interface (if it same interface type- IPv4)
For example: Interface Voice
VLAN ID 1 Group 1
Interface External VLANID 10 Group 2
Make sure that the ports allocated in the’ Physical port setting’ have corresponding VLAN ID
For example Group 1 VLANID 1
Group 2 VLANID 10
Otherwise the packets will be tagged with the VLAN ID specified in the multiply interface table
2. Save the settings to flash memory ("burn") and reset the device.
4.3.3. ENABLE THE SBC APPLICATION
The procedure below describes how to enable the E-SBC application. Once enabled, the E-SBC specific parameters/pages become available in the W eb interface.
 To enable SBC :
1.
Open the 'Applications Enabling' page (Configuration tab > VoIP menu > Applications
Enabling submenu > Applications Enabling), and then from the 'SBC' drop-down list, select
'Enable':
Figure 4.3-3: Enabling SBC Application
PAGE 27 OF 45
2.
Click Submit.
3.
Save the settings to flash memory ("burn") and reset the device.
4.3.4. DEFINE NUMBER OF MEDIA CHANNELS
If transcoding is required, the number of DSP channels must be defined, as described in the
procedure below. Note that each SBC session comprises two legs. Therefore, if you enter 240, a
maximum of 120 calls can occur simultaneously.
 To define the number of media channels for transcoding:
1.
Open the 'IPmedia Parameters' page (Configuration tab > VoIP menu > Media submenu >
IPMedia Settings), and then in the 'Number of Media Channels' field, enter the number of SBC
legs that require transcoding.
Figure 4.3-4: Defining Number of SBC Sessions
2.
3.
Click Submit.
4.3.5. RESET AUDIOCODES ESBC
 To reset AudioCodes ESBC :
1.
Open ‘Maintenance Actions’ page (‘Maintenance’ tab > Maintenance > Maintenance Actions)
then select Reset > note ‘burn to FLASH’ option is enable by default
Figure 4.3-5: Reset AudioCodes device
PAGE 28 OF 45
4.3.6. CONFIGURE SIP AND RTP INTERFACES
The procedure below describes how to configure SIP signaling interfaces and RTP (media)
interfaces. The SIP signaling interfaces are defined as SIP Interfaces; the RTP interfaces are
defined as Media Realms. These are associated together under one entity termed SRD (Signaling
Routing Domain). In the scenario, you need to define the SRD as follows:
 LAN SRD: with SIP signaling interface Voice port 5060 and RTP traffic port range 6000-6240.

WAN SRD: with SIP signaling interface External port 5060 and RTP traffic port range 70007240.
 To configure SIP and RTP interfaces:
1.
Configure a Media Realm for the RTP traffic in the 'SIP Media Realm Table' page
(Configuration tab > VoIP menu > Media submenu >Media Realm Configuration):
•
Add the Media Realm entry for the E-SBC LAN network signaling leg:
a. Media Realm Name = MR_PBX
b. IPv4 Interface Name = Voice
c. IPv6 Interface Name = None
d. Port Range Start = 6000
e. Number of Media Session Legs = 25
f.
Trans Rate Ratio = 0
g. Is Default = Yes
h. Click Submit; the Port Range End is automatically calculated. For example, the first
session uses port 6000, the second session port 6010, and so on.
Figure 4.3-6: Defining LAN Media Realm
•
PAGE 29 OF 45
Add the Media Realm entry for the E-SBC W AN network signaling leg:
a. Media Realm Name = MR_W AN
b. IPv4 Interface Name = External
c. IPv6 Interface Name = None
d. Port Range Start = 7000
e. Number of Media Session Legs = 25
f.
g.
h.
Trans Rate Ratio = 0
Is Default = No
Click Submit; the Port Range End is automatically calculated. For example, the first
session uses port 7000, the second session port 7010, and so on.
Figure 4.3-7: Defining W AN Media Realm
2.
Reset AudioCodes ESBC
3.
Configure an LAN SRD in the 'SRD Table' page (Configuration tab > VoIP menu > Control
Network submenu > SRD Table):
•
Add the SRD for the network signaling leg:
a. Name = PBX
b. Media Realm = MR_PBX (Note that this string must be identical to the Media Realm
Name string defined in the SIP Media Realm table)
c. Click Apply.
Figure 4.3-8: Defining LAN SRD
PAGE 30 OF 45
4.
Configure an W AN SRD in the 'SRD Table' page (Configuration tab > VoIP menu > Control
Network submenu > SRD Table):
a. Add the SRD for the network signaling leg:
b. Name = W AN_SRD
c. Media Realm = MR_W AN (Note that this string must be identical to the Media Realm
Name string defined in the SIP Media Realm table)
d. Click Apply.
Figure 4.3-9: Defining W AN SRD
5.
Configure a SIP signaling interfaces in the 'SIP Interface Table' page (Configuration tab >
VoIP menu > Control Network submenu > SIP Interface Table):
•
•
Add the LAN SIP interface for the network signaling leg:
a. Network Interface = Voice (Note that this string must be identical to the Interface
Name string defined in the Multiple Interface table)
b. Application Type = SBC
c. UDP Port, TCP Port, TLS Port = enter the values 5060, 5060, and 5061 respectively.
d. SRD = 1 (Note: This is the SRD index defined in the SRD table)
e. Click Apply.
Add the W AN SIP interface for the network signaling leg:
a. Network Interface = External (Note that this string must be identical to the Interface
Name string defined in the Multiple Interface table)
b. Application Type = SBC
c. UDP Port, TCP Port, TLS Port = enter the values 5060, 5060, and 5061 respectively.
d. SRD = 2 (Note: This is the SRD index defined in the SRD table)
e. Click Apply.
Figure 4.3-10: Defining the SIP Interfaces
PAGE 31 OF 45
4.3.7. DEFINE PROXY SETS
The procedure below describes how to configure Proxy Sets. The Proxy Set defines the IP
addresses of the ShoreTel IP PBX and Telstra SIP Connect servers.
 To configure Proxy Sets:
1.
2.
Open the 'Proxy Sets Table' page (Configuration tab > VoIP menu > Control Network
submenu > Proxy Sets Table).
Add a Proxy Set for ShoreTel
3.
trunk switch configured in
a. Proxy Set ID = 1
step 4.2.3
b. Proxy Address = 10.8.53.16
c. Transport Type = UDP
d. SRD Index = 1 (Note that this is the SRD index previously defined for the E-SBC )
e. Click Submit.
Add a Proxy Set for Telstra SIP Connect
IP address of ShoreTel SIP
a.
b.
c.
d.
e.
FQDN of Telstra
Proxy Set ID = 2
Proxy – choose
correct Proxy for
Proxy Address = sbc-cw.ipvs.net
your state
Transport Type = UDP
SRD Index = 2 (Note that this is the SRD index previously defined for the E-SBC )
Click Submit.
Figure 4.3-11: Defining Proxy Set for ShoreTel
PAGE 32 OF 45
Figure 4.3-12: Defining Proxy Set for Telstra
4.
4.3.8. CREATE IP PROFILES
The procedure below describes how to configure IP profiles that can be assigned to different IP
groups allowing setting of specific parameters per IP group to configure IP Profiles ID 1 and 2
1.
2.
Open the 'Proxy Sets Table' page (Configuration tab > VoIP menu > Coders and Profiles >
IP Profiles Setting).
Create IP profile ID 1
a.
Profile name: PBX
b.
Set Disconnect on broken connection to “yes”
c.
SBC Remote Update Support to ‘Not supported’
d.
SBC Remote Re-Invite Support to ‘Not supported’
NOTE:
•
Item b is required to be set up to prevent E-SBC
performs transfer – no RTP is being send
to disconnect the call if ShoreTel
•
3.
Item c and d are required to handle correct transfer from ShoreTel to TIPIT extension and
back to ShoreTel.( if this parameters are not set no audio issue may be experienced if
TIPIT extension transfers back to ShoreTel) Both parameters force E-SBC to handle all
incoming re-invites from Telstra and not forward to ShoreTel
Create IP profile ID 2
a. Profile name Telstra
b.
PAGE 33 OF 45
Set Disconnect on broken connection to “yes”
Figure 4.3-13: Creation IP Profile ID 1
Figure 4.3-14: Creation IP Profile ID 2
PAGE 34 OF 45
4.3.9. DEFINE IP GROUPS
An IP Group is a convenient way to represent a SIP User Agent (client or server) entity, which in our
network are the ShoreTel and Telstra. An IP Group is defined with a set of characteristics, such as
with an SRD and an IP Profile. In our scenario, we need to define two IP Groups.
 To configure IP Groups:
1.
2.
Open the 'IP Group Table' page (Configuration tab > VoIP menu > Control Network
submenu > IP Group Table).
Add an IP Group for ShoreTel with the following values:
3.
a. Index = 1
b. Type = SERVER
c. Description = PBX
d. Proxy Set ID = 1
e. SIP Group Name = 10.8.53.16
f.
SRD : 1
g. Media Realm = MR_PBX
h. IP Profile: 1
Click Submit.
Figure 4.3-15: Defining IP Group for ShoreTel
4.
5.
PAGE 35 OF 45
Add an IP Group for the Telstra with the following values:
a. Index = 2
b. Type = SERVER
c. Description = Telstra
d. Proxy Set ID = 2 (as defined previously, which shows the IP address/FQDN of Telstra Sip
Connect )
e. SIP Group Name = sipconn.test1.com
SIP Realm/Domain
f.
Contact User : 353950600
from your RASS
g. SRD : 2
Order
h. Media Realm =MR-W AN
Trunk Pilot Number
from your RASS
i.
IP Profile: 2
Order (without
leading ‘0’)
Click Submit.
Figure 4.3-16: Defining IP Group for Telstra
6.
4.3.10. CONFIGURE ACCOUNT TABLE FOR TELSTRA SIP CONNECT REGISTRATION
An account must be configured on the AudioCodes ESBC in order for it to register and/or
authenticate to the Telstra SIP Connect service. This will allow the AudioCodes ESBC to place
outgoing calls and receive incoming calls on behalf of the ShoreTel. Open the ‘Account Table’ page
(Configuration tab > VoIP menu > SIP Definitions submenu > Account Table).
1.
Enter “1” in the text box and click “Add”
2.
Enter the following values accordingly:
a.
b.
c.
d.
e.
f.
g.
h.
i.
j.
Enter information
from your RASS
Order
[Index 1]
Served Trunk Group = -1
Served IP Group = 1
Serving IP Group = 2
Username = (as per Telstra Provisioning information) e.g. N3301011R
Password = (as per Telstra Provisioning information )
Host Name = (as per Telstra Provisioning information) e.g. sipconn.test1.com
Register = Yes
Contact User = (Pilot number) e.g. 353950600
Application Type = SBC
Trunk Pilot Number
Figure 4.3-17: Defining Account Table
from your RASS
Order (without
leading ‘0’)
4.3.11. DEFINE IP-TO-IP ROUTING RULES
The procedure below describes how to configure IP-to-IP routing rules for routing SIP signaling and
calls between IP Groups.
PAGE 36 OF 45
 To configure IP-to-IP routing rules:
1.
2.
Open the 'IP2IP Routing Table' page (Configuration tab > VoIP menu > SBC submenu >
Routing SBC submenu > IP to IP Routing Table).
Add a routing rule to route calls from IP Group 1 (ShoreTel) to IP Group 2 (Telstra):
[Index 1] (Highlighted values are non-default)
a.Source IP Group ID = 1
b. Source Username Prefix = *
c. Source Host = *
d. Destination Username Prefix = *
e. Destination Host = *
f.
Request Type = All
g. Message Condition = None
h. Destination Type = IP Group
i.
Destination IP Group ID = 2
j.
Destination SRD = 2
k. Destination Address = (empty)
l.
Destination Port = 0
m. Destination Transport Type = (empty) (default UDP)
n. Alternative Route Options = Route Row
o. Cost Group = None
3.
Click Submit
4.
Add a routing rule to route calls from IP Group 2 (Telstra SIP Connect ) to IP Group 1
(ShoreTel )
[Index 2] (Highlighted values are non-default)
a.Source IP Group ID = 2
b. Source Username Prefix = *
c. Source Host = *
d. Destination Username Prefix = *
e. Destination Host = *
f.
Request Type = All
g. Message Condition = None
h. Destination Type = IP Group
i.
Destination IP Group ID = 1
j.
Destination SRD = 1
k. Destination Address = (empty)
l.
Destination Port = 0
m. Destination Transport Type = (empty) (default UDP)
n. Alternative Route Options = Route Row
o. Cost Group = None
5.
Click Submit
Figure 4.3-18: IP-to-IP Routing Rules between IP Groups
PAGE 37 OF 45
4.3.12. DEFINE MANIPULATION RULES
 Outbound manipulation rules
1. Open the 'IP to IP Outbound Manipulation Table' page (Configuration tab > VoIP menu >
SBC submenu > Manipulation SBC submenu > IP to IP Outbound Manipulation Table).
2. Create the flowing rules:
a. Index 0: remove 3 digits from destination number received from ShoreTel starting with
‘+61’
b. Index 1: remove 1 digit from the destination number and add 0011 received from ShoreTel
starting with ‘+’
c. Index 3: remove 3 digits from the source number received from ShoreTel ( this will be the
range allocated to Telstra services )
d. Index 4: remove 3 digits from the source number received from ShoreTel ( this will be the
range allocated to Telstra services )
e. Index 6: for all other source numbers recede from ShoreTel leave 0 from the right and add
pilot number e.g. 353950600 – this rule will alternate source number if it is not part of the
Telstra allocated range ( in case of call forward for example)
f.
Index 7: all four digits source numbers received from Telstra starting with the last 4 digits of
DID will have the full number appended (This is implemented due to the fact that if two
ShoreTel extension where calling each other via DID, incoming invite from Telstra would
include only 4 digits in the TO header. For example if 53950601 calling 53950602 the
incoming invite from Telstra would include only 0601 in the source number. )
g. Index 8: - same as index 7
h. Index 9: all destination numbers coming from Telstra remove 1 from the left to send only 8
digit number to ShoreTel
Rules like index 3 & 7 in the table below need to be added for each DID range on the SIP Service
(consult your RASS Order)
PAGE 38 OF 45
Table 1-Highlighted cells are non-default, rows in italics require info from RASS Order
Is
Additio Src Dest
Src
Ind
nal
IPGro IPGro Username
ex
Prefix
Manipul up ID up ID
ation
Lea
Dest
Re
Remo
ve
Privacy
Src
Dest Requ
Manip
Remov
Suffi
Fro Prefix2Ad
Userna
Route Trigge
ve
Restrict
Hos
Hos est
ulated
e From
x2A
me
IPGro
r
From
m
d
ion
t
t
Type
URI
Right
dd
Prefix
up ID
Left
Rig
Mode
ht
0
0
1
2
"*"
"*"
+61
"*"
0 (All) -1
0
(Any)
1
(Desti 3
nation)
0
255 ""
""
0
(Transp
arent)
1
0
1
2
"*"
"*"
+
"*"
0 (All) -1
0
(Any)
1
(Desti 1
nation)
0
255 0011
""
0
(Transp
arent)
3
0
1
2
+613539506
"*"
[00-04]
"*"
"*"
0 (All) -1
0
(Any)
0
(Sourc 3
e)
0
255 ""
""
0
(Transp
arent)
4
0
1
2
+613539509
"*"
[20-24]
"*"
"*"
0 (All) -1
0
(Any)
0
(Sourc 3
e)
0
255 ""
""
0
(Transp
arent)
6
0
1
2
"*"
"*"
"*"
"*"
0 (All) -1
0
(Any)
0
(Sourc 0
e)
0
0
353950600 ""
0
(Transp
arent)
7
0
2
1
09[20-24]#
"*"
"*"
"*"
0 (All) -1
0
(Any)
0
(Sourc 0
e)
0
255 035395
""
0
(Transp
arent)
8
0
2
1
06[00-04]#
"*"
"*"
"*"
0 (All) -1
0
(Any)
0
(Sourc 0
e)
0
255 035395
""
0
(Transp
arent)
9
0
2
1
"*"
"*"
"*"
"*"
0 (All) -1
0
(Any)
1
(Desti 1
nation)
0
255 ""
""
0
(Transp
arent)
Figure 4.3-19: IP-to-IP Outbound Manipulation
 Message Manipulation rules
1.
Open the 'Message Manipulation Table' page (Configuration tab > VoIP menu > SIP
Definitions submenu > Msg Policy & Manipulation submenu > Message Manipulation
Table).
Set ID 2 will be assigned to Outbound Manipulation for IP group 2
2. Create the following rules :
a. Index 1: – all Invites will have replaced host name with 'sipconn.test1.com' in the URL of the
From header
PAGE 39 OF 45
b.
c.
d.
Index 3 - all reinvites to Telstra will have replaced a=send only with a=sendrcv – this is to
allow ShoreTel to Play MOH to external party
Index 4 : - all invites to Telstra will have replaced the host URL in P-asserted identity
header with the host URL in TO header (this is to replace ShoreTel sending P-asserted :
[email protected] with [email protected]
Index: 7 – removes certain headers starting with “media-release” from reinvite being sent to
Telstra. (Not removing these headers caused no Audio in certain call forward scenarios)
SIP Realm/Domain
from your RASS
Order
2 - Copy and Paste these values to your configuration, except as noted.
Ma
In
n
de
Set
x
ID
Message Type
Condition
Action Subject
header.from.url.host
Action
Type
Action
Value
Row Role
0 (Use
2 (Modify) 'sipconn.te Current
st1.com'
Condition)
1
2
Invite
3
2
ReInvite.Request
4
2
invite
header.p-assertedidentity.url.host
2 (Modify)
7
2
body.sdp regex (.*)(a=mediaReinvite.Request release:.*)(a=mediarelease.*)(m=audio.*)(.*)
body.sdp
0 (Use
2 (Modify) $1+$4+$5 Current
Condition)
0 (Use
param.message.sdp.rtpmode==' param.message.sdp.r
2 (Modify) 'sendrecv' Current
sendonly'
tpmode
Condition)
0 (Use
header.to.
Current
url.host
Condition)
Figure 4.3-20 - Message Manipulation Table
3. Apply Manipulation Set 2 to Outbound manipulation in IP group 2
a. Navigate to VoIP -> Control Network -> IP Group Table.
b. Select IP Group 2.
c. Set “Outbound Message Manipulation Set” to 2.
PAGE 40 OF 45
Figure 4.3-21: Add Message manipulation to IP group
4.3.13. E-SBC SECURITY
This document doesn’t cover additional security options on ESBC. If advanced security is
required refer to LTRT-30200_Recommended_Security_Guidelines_Technical_Note
PAGE 41 OF 45
5.
VENDOR SUPPORT
ShoreTel Technical Support Number: +61 2 9959 8080.
6.
REFERENCES
DOCUMENT NUMBER
7.
TITLE
DEFINITIONS
TERM
DEFINITION
AS
Application Server – BroadWorks in the case of TIPT
DNS
Domain Name Server
DTMF
Dual Tone Multi-Frequency
PSTN
Public Switched Telephone Network
RASS
Record Automation for Special Services
SBC
Acme Packet – Session Border Controller
(i.e. VoIP Firewall used to provide NAT and demarcation)
SIP
Session Initiation Protocol
SIP Trunk
A logical interface which delivers PSTN access via TIPT voice
channels to the customers PBX
TIPT
Telstra IP Telephony
Trunk Group
A collection of TIPT numbers and services within a BW Group used
to allocate the SIP trunk to the customer site
VOIP
Voice-over-IP
8.
ATTACHMENTS
DOCUMENT NUMBER
TITLE
PAGE 42 OF 45
9.
APENDIX A – AUDIOCODES M1000 MSBG INTERFACE SETUP
9.1.1. DEFINE IP ADDRESSES FOR OAMP, MEDIA AND CONTROL
The procedure below describes how to configure the IP interface for each network.
 To configure the LAN IP interface:
1.
2.
Open the Multiple Interface table (Configuration tab > VoIP menu > Network submenu > IP
Settings). Assign the Application Type, IP address, Prefix Length/Subnet Mask, Gateway,
VLAN ID, Interface Name and DNS Server. Take note that for the Interface Name the word
“W AN” should NOT be used as it is a reserved word. The Interface Name can be chosen
arbitrarily.
Add the IP interface entry for the network signaling leg:
a. Application Type = OAMP + Media + Control
b. IP address = 10.99.0.20
c. Prefix Length = 24
d. Gateway = 10.99.0.21
e. VLAN ID = 1
f. Interface Name = Voice
g. Primary DNS Server IP Address = 10.86.113.20
h. Secondary DNS Server IP Address = 10.86.114.20
i. Click Apply, and then done to validate the entry.
Defining the IP Address
3.
 To configure the LAN Switch Vlan 1 interface:
This interface is will route between LAN and WAN interfaces: The IP address assigned
should be the same as the IP address assigned to gateway filed in the ‘Multiply Interface’
Table
4. Open connection page (Configuration >>Data>Data System >Connections) Select edit icon
next to ‘LAN switch Vlan1’ parameter, select Setting tab then enter the IP address
PAGE 43 OF 45
Defining the Internal LAN switch Vlan 1 IP Address
 To configure the WAN interface:
5.
Open ‘W AN access setting Page‘(Configuration > Data > W AN Access > Setting) and
configure the flowing
a. Connection Type: Manual IP address Connection
b.
c.
IP address: 10.2.1.210
Netmask : 255.255.255.0
d.
Default gateway: 10.2.1.1
e.
Primary DNS : If required
Defining the WAN IP Address
•
Proceed with the steps defined in the main document section 4.3.3
•
In sections 4.3.4 – 4.3.12 all reference to the “External” interface should be replaced with the
“W AN” interface created in this appendix.
PAGE 44 OF 45
10. DOCUMENT CONTROL SHEET
Contact for Enquiries and Proposed Changes
If you have any questions regarding this document contact:
NAME:
LEN BANKS
DESIGNATION:
Technology Specialist / CPE
Accreditation
PHONE:
+61 3 8647 9846
FAX:
+61 3 9650 1756
EMAIL:
[email protected]
If you have a sugge sti on for i mproving thi s document, pl ease contact the pe rson l i sted above .
ISSUE NO.
ISSUE DATE
NATURE OF AMENDMENT
0.1
01/05/2013
Initial release for comment
1.0
24/05/2013
First approved release
PAGE 45 OF 45

SIP Connect for ShoreTel UC with AudioCodes ESBC Customer

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