ADTRAN SBC and ShoreTel SIP Trunk Interoperability

ADTRAN SBC and ShoreTel SIP Trunk Interoperability
6AOSSG0009-42B
May 2014
Interoperability Guide
ADTRAN SBC and ShoreTel SIP Trunk
Interoperability
This interoperability guide provides instructions for integrating an
ADTRAN session border controller (SBC) and a ShoreTel® IP phone
system using a Session Initiation Protocol (SIP) trunk to provide a
connection to the service provider network. This guide includes the
description of the network application, verification summary, and
individual device configurations for the ADTRAN SBC and the ShoreTel
products.
For additional information on configuration of the ADTRAN products,
please visit the ADTRAN Support Community at
https://supportforums.adtran.com
This guide consists of the following sections:
•
•
•
•
•
•
•
•
•
Overview on page 2
Hardware and Software Requirements and Limitations on page 2
Verification Performed on page 3
ShoreTel Features and Exceptions on page 6
Configuring the ADTRAN SBC on page 6
ADTRAN SBC Sample Configuration on page 12
Configuring the ShoreTel PBX on page 13
Troubleshooting on page 25
Additional Resources on page 26
SBC and ShoreTel Interoperability
Overview
Overview
Service providers are increasingly using SIP trunks to provide Voice over IP (VoIP) services to customers.
ADTRAN SBCs terminate the SIP trunk from the service provider and interoperate with the customer's IP
private branch exchange (PBX) system. A second SIP trunk from the gateway connects to the IP PBX. The
ADTRAN SBC operates as a SIP back-to-back user agent (B2BUA) and acts as a gateway to the service
provider for SIP trunking. The ADTRAN SBC features normalize the SIP signaling and media between the
service provider and the customer IP PBX. Figure 1 illustrates the use of the ADTRAN SBC in a typical
network deployment.
Additional information is available online at ADTRAN’s Support Community,
https://supportforums.adtran.com. Specific resources are listed in Additional Resources on page 26.
Internet SIP Trunking
Service Provider
NAT Firewall
Da
Enterprise
LAN
ta
Data
Voic
e
OK
ITSP
Adtran SBC
®
MENU
MENU
PSTN
Internet
192.168.1.0
e
Voic
CANCEL
3
1
2
4
5
7
8
9
*
0
#
6
SIP Server
IP Phone
ShoreTel ShoreGear
ShoreTel Call Manager/Director
Figure 1. ADTRAN SBC in the Network
Hardware and Software Requirements and Limitations
Interoperability with the ShoreTel IP phone system is available on ADTRAN products with the SBC
feature code as outlined in the AOS Feature Matrix, available online at ADTRAN’s Support Forum,
https://supportforums.adtran.com. The test equipment, testing parameters, and associated caveats are
described in the following sections.
Equipment and Versions
Products are certified via the Technology Partner Validation Process for the ShoreTel IP phone system. The
table below lists the firmware releases certified for both the AOS SBC and the ShoreTel IP phone system.
Table 1. Verification Test Firmware Versions
Product
6AOSSG0009-42B
Firmware Version
AOS SBC
R10.5.3
ShoreTel
14.1
Copyright © 2014 ADTRAN, Inc.
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SBC and ShoreTel Interoperability
Verification Performed
Verification Performed
Interoperability verification testing focused on SIP trunk operations between the ADTRAN SBC IP
business gateway and the ShoreTel IP phone system. PBX features not specific to basic SIP trunking were
not included in this verification.
Table 2. Initialization and Basic Calls
ID
Name
Description
Notes
1.1
Setup and Initialization
Verify successful set up and initialization of the system
under test (SUT).
PASS
1.2
Outbound Call
(Domestic)
Verify calls outbound placed through the SUT reach the
external destination.
PASS
1.3
Inbound Call (Domestic)
Verify calls received by the SUT are routed to the default
trunk group destination.
PASS
1.4
Device Restart – Power
Loss
Verify that the SUT recovers after power loss to the SUT.
PASS
1.5
Device Restart – Network
Loss
Verify the SUT recovers after loss of network link to the
SUT.
PASS
1.6
All Trunks Busy –
Inbound Callers
Verify an inbound caller hears the busy tone when all
channels/trunks are in use.
PASS
1.7
All Trunks Busy –
Outbound Callers
Verify an outbound caller hears the busy tone when all
channels/trunks are in use.
PASS
1.8
Incomplete Inbound Calls
Verify proper call progress tones are provided and
proper call teardown for incomplete inbound calls.
PASS
Table 3. Media and Dual-Tone Multi-Frequency (DTMF) Support
ID
Name
Description
Notes
2.1
Media Support –
ShoreTel Phone to SUT
Verify call connection and audio path from a ShoreTel
phone to an external destination through the service
provider using all supported coder-decoders (CODECs)
with both sides set to a common CODEC.
PASS
2.1
Media Support – SIP
Reference to SUT
Verify call connection and audio path from SIP reference
phones to an external destination through the service
provider using all supported CODECs with both sides
set to a common CODEC.
PASS
2.2
CODEC Negotiation
Verify CODEC negotiation between the SUT and the
calling device with each side configured for a different
CODEC.
PASS
2.3
DTMF Transmission –
Out-of-Band/Inband
Verify transmission of inband and out-of-band digits per
RFC 2833 for various devices connected to the SUT.
PASS
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SBC and ShoreTel Interoperability
Verification Performed
Table 3. Media and Dual-Tone Multi-Frequency (DTMF) Support (Continued)
2.4
Auto Attendant Menu
Verify that inbound calls are properly terminated on the
ShoreTel Auto Attendant menu and that calls can be
transferred to the desired extension.
PASS
2.5
Auto Attendant Menu Dial by Name
Verify that inbound calls are properly terminated on the
ShoreTel Auto Attendant menu and that calls can be
transferred to the desired extension using the Dial by
Name feature.
PASS
2.6
Auto Attendant Menu
Checking Voice Mailbox
Verify that inbound calls are properly terminated on the
ShoreTel Auto Attendant menu and calls can be
transferred to the Voicemail Login Extension.
PASS
Table 4. Performance and Quality of Service
ID
Name
Description
Notes
3.1
Voice Quality Service
Levels
Verify the SUT can provide a voice quality service level
agreement (SLA) across the WAN from the customer
premises to the SUT SIP gateway.
PASS
3.2
Capacity Test
Verify the service provider interface can sustain services
through periods of heavy outbound and inbound load.
PASS
3.3
Post Dial Delay
Verify that the post dial delay is within acceptable limits.
PASS
3.4
Billing Accuracy
Verify that all test calls made are accurately reflected in
the SUT’s Call Detail Record (CDR) and billing reports.
PASS
Table 5. Enhanced Services and Features
ID
Name
Description
Notes
4.1
Caller ID Name and
Number - Inbound
Verify that the caller ID name and number are received
from the SIP endpoint device.
PASS
4.2
Caller ID Name and
Number - Outbound
Verify that caller ID name and number are sent from
SIP endpoint device.
PASS
4.3
Hold from SUT to SIP
Reference
Verify connected call can be successfully put on hold
and resumed.
PASS
4.4
Call Forward - SUT
Verify outbound calls that are being forwarded by the
SUT are redirected and connected to the appropriate
destination.
PASS
4.5
Call Transfer - Blind
Verify a call connected from the SUT to the ShoreTel
phone can be blind transferred to an alternate
destination.
PASS
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SBC and ShoreTel Interoperability
Verification Performed
Table 5. Enhanced Services and Features (Continued)
4.6
Call Transfer Consultative
Verify a call connected from the SUT to the ShoreTel
phone can be transferred with consultation to an
alternate destination.
PASS
4.7
Conference - Ad Hoc
Verify successful ad hoc conference of three parties.
PASS
4.8
Inbound Direct Inward
Dialing/Dialed Number
Identification Service
(DID/DNIS)
Verify the SUT provides inbound dialed number
information and is correctly routed to the configured
destination.
PASS
4.9
Outbound 911
Verify that outbound calls to 911 are routed to the
correct Public Safety Answering Point (PSAP) for the
calling location and that caller ID information is
delivered.
Note: If 9911 is dialed, the call goes out immediately. If
911 is dialed, there is a 5 second delay. This is
documented in the ShoreTel System Administration
Guide. “If the user forgets to dial an access code
before dialing 911, the system waits five seconds
before routing the call to a 911-capable trunk. This
pause has been introduced to eliminate accidental calls
to 911.”
Conditional
Pass
Note: This
was tested
in a
controlled
environment
without
actual
circuits, just
to verify call
placement.
4.10
Operator Assisted
Verify that 0+ calls are routed to an operator for calling
assistance.
PASS
4.11
Inbound/Outbound
Call with Blocked
Caller ID
Verify that calls with blocked caller ID route properly,
and the answering phone does not display any caller
ID information.
PASS
4.12
Inbound Call to a Hunt
Group
Verify that calls route to the proper hunt group and are
answered by an available hunt group member with
audio in both directions using G.729 and G.711
CODECs.
PASS
4.13
Inbound Call to a
Workgroup
Verify that calls route to the proper workgroup and are
answered successfully by an available workgroup
agent with audio in both directions using G.729 and
G.711 CODECs.
PASS
4.14
Inbound Call to
DNIS/DID and Leave a
Voicemail Message
Verify that inbound calls to a user, via DID/DNIS, routes
to the proper user mailbox and a message can be left
with proper audio.
PASS
4.15
Call Forward - FindMe
Verify that inbound calls are forwarded to a user’s
FindMe destination.
PASS
4.16
Call Forward Always
Verify that inbound calls are immediately and
automatically forwarded to a user’s external
destination.
PASS
4.17
Inbound/Outbound
Fax Calls
Verify that inbound/outbound fax calls complete
successfully.
PASS
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SBC and ShoreTel Interoperability
ShoreTel Features and Exceptions
Table 5. Enhanced Services and Features (Continued)
4.18
ShoreTel UCB
Verify that inbound calls are properly forwarded to the
ShoreTel UCB, that it properly accepts the access
code, and that the conference bridge functions
properly.
PASS
4.19
Inbound Call to
Bridged Call
Appearance (BCA)
Extension
Verify that inbound calls are properly presented to all
phones that have BCA configured and that the call can
be answered, placed on hold, and then transferred.
PASS
4.20
Inbound Call to a
Pickup Group
Verify that inbound calls are properly presented to all of
the phones that have a pickup group configured and
that the call can be answered, placed on hold, and then
transferred.
PASS
Table 6. Security
ID
5.1
Name
Digest Authentication
Description
Verify the SUT supports the use of digest
authentication for service access for both inbound and
outbound calls.
Notes
N/A
ShoreTel Features and Exceptions
The configuration information provided in this document offers example configurations for the ShoreTel
IP phone system and the ADTRAN SBC unit. Even though configuration requirements can vary among
installations, the information provided in these steps, along with documentation provided by ADTRAN
and the service provider, should prove to be sufficient. However, every design can vary and some may
require more planning than others. Refer to the ShoreTel Planning and Installation Guide, available from
the ShoreTel support website: http://support.shoretel.com/.
ShoreTel Supported Features
For more information on ShoreTel supported features of SIP trunks, refer to the ShoreTel Administration
Guide available from the ShoreTel support website: http://support.shoretel.com/.
Some feature limitations may occur when using SIP trunks. For more information on feature limitations,
refer to the ShoreTel Administration Guide and ShoreTel Partner Guide, available from the ShoreTel
support website: http://support.shoretel.com/.
Configuring the ADTRAN SBC
The SBC can be configured using either the AOS command line interface (CLI) or the web-based
graphical user interface (GUI). The following sections describe the key configuration settings required for
this solution using the CLI.
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SBC and ShoreTel Interoperability
Configuring the ADTRAN SBC
To configure the SBC for interoperability with the ShoreTel IP phone system, follow these steps:
•
•
•
•
•
•
•
•
•
Step 1: Accessing the SBC CLI on page 7
Step 2: Configuring the Basic Network Settings on page 8
Step 3: Configuring Global Voice Modes for Local Handling on page 8
Step 4: Enabling Media Anchoring on page 8
Step 5: Configuring Header Manipulation Rules on page 9
Step 6: Configuring the Service Provider SIP Trunk on page 9
Step 7: Configuring the ShoreTel PBX SIP Trunk on page 10
Step 8: Configuring a Trunk Group for the Service Provider on page 10
Step 9: Configuring a Trunk Group for the ShoreTel PBX on page 11
Step 1: Accessing the SBC CLI
The AOS unit can be managed by console port, http, https, telnet and SSH. Most of the initial
configuration is performed through the console port or Telnet session. Accessing the AOS unit is described
in this step.
To access the CLI on your AOS unit, follow these steps:
1. Boot up the unit.
2. Telnet to the unit (telnet <ip address>), for example:
telnet 10.10.10.1.
If during the unit’s setup process you have changed the default IP address (10.10.10.1),
use the configured IP address.
3. Enter your user name and password at the prompt.
The AOS default user name is admin and the default password is password. The default
enable password is password. If your product no longer has the default user name and
passwords, contact your system administrator for the appropriate user name and
passwords.
4. Enable your unit by entering enable at the prompt as follows:
>enable
5. If configured, enter your Enable mode password at the prompt.
6. Enter the unit’s Global Configuration mode as follows:
#configure terminal
(config)#
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SBC and ShoreTel Interoperability
Configuring the ADTRAN SBC
Step 2: Configuring the Basic Network Settings
Basic network configuration includes setting up two Ethernet interfaces, one for the Ethernet LAN
interface to the ShoreTel unit, and the second for the Ethernet WAN interface to the service provider. Both
interfaces are configured using the ip address <ipv4 address> <subnet mask> and media-gateway ip
primary commands. The ip address command configures a static IP address for the interface, and the
media-gateway command is required on the interface for SIP and Realtime Transport Protocol (RTP)
media traffic.
Enter the commands from the Ethernet interface configuration mode as follows:
For the LAN:
(config)#interface ethernet 0/1
(config-eth 0/1)#description LAN
(config-eth 0/1)#ip address 192.168.1.1 255.255.255.0
(config-eth 0/1)#media-gateway ip primary
(config-eth 0/1)#no shutdown
For the WAN:
(config)#interface ethernet 0/2
(config-eth 0/2)#description WAN
(config-eth 0/2)#ip address 203.0.113.2 255.255.255.252
(config-eth 0/2)#media-gateway ip primary
(config-eth 0/2)#no shutdown
Step 3: Configuring Global Voice Modes for Local Handling
Configure the ADTRAN SBC to use the local mode for call forwarding and transfer handling. By default,
both of these functions are handled by the network. To change these settings, use the voice transfer-mode
local and voice forward-mode local commands. Enter these commands from the Global Configuration
mode. By using the local parameter, both commands specify allowing the unit to handle call forwarding
and transfers locally.
Enter the commands as follows:
(config)#voice transfer-mode local
(config)#voice forward-mode local
Step 4: Enabling Media Anchoring
Media anchoring is an SBC feature that routes RTP traffic through the ADTRAN SBC gateway. Minimum
configuration for media anchoring includes enabling the feature using the ip rtp media-anchoring
command from the Global Configuration mode. The RTP symmetric filter works in conjunction with
media anchoring to filter nonsymmetric RTP packets. Enable the RTP symmetric filter using the ip rtp
symmetric-filter command. Enter the commands as follows:
(config)#ip rtp media-anchoring
(config)#ip rtp symmetric-filter
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SBC and ShoreTel Interoperability
Configuring the ADTRAN SBC
Step 5: Configuring Header Manipulation Rules
The following configuration example is specific to the ShoreTel IP phone system. For a generic example of
a basic SBC configuration, refer to SBC SIP Trunking Sample Configuration.
A header manipulation rule (HMR) is created to convert a 183 Session Progress with Session Description
Protocol (SDP) to a 180 Ringing with SDP. The ShoreTel unit expects a 180 Ringing, but some service
providers send a 183 Session Progress instead. If your service provider never sends a 183 Session Progress,
then the Shoretel_Early_Media_Workaround rule set and the Shoretel_Outbound HMR policy are
unnecessary. However, they will not negatively affect the operation of the SBC if they are configured and
not needed.
Refer to Manipulating SIP Headers and Messages in AOS for a more complete discussion of SIP HMR
policies and rule sets.
Enter the commands as follows:
(config)#hmr policy ShoreTel_Outbound
(config-policy-ShorTel_Outbound)#rule-set ShoreTel_Early_Media_Workaround 10
!
(config)#hmr rule-set ShoreTel_Early_Media_Workaround
(config-rule-set-ShoreTel_Early_Media_Workaround)#message-rule Convert_183_to_180
message-type response 10
(config-msg-rule-Convert_183_to_180)#modify header sip-status-line position first match-value "/183
Session Progress/" new-value "/180 Ringing/" 10
Step 6: Configuring the Service Provider SIP Trunk
The first of two voice trunks that must be configured is the SIP trunk to the service provider from the
ADTRAN SBC. Check with your service provider for any specific requirements beyond those listed in this
document. Your service provider will provide you with the IP addresses or FQDN and possibly the port
numbers for their SIP server. They may also provide a backup or secondary SIP server. This example uses
one SIP server with the default port number (5060).
Use the voice trunk <Txx> type sip command to define a new SIP trunk and activate the Voice Trunk
Configuration mode for the individual trunk. From the Voice Trunk Configuration mode, you can provide a
descriptive name for the trunk and define the SIP server’s primary IPv4 address (or host name). Use the
description <text> command to label the trunk. Use the sip-server primary <ipv4 address | hostname>
command to define the host name or IPv4 address of the primary server to which the trunk sends SIP
messages.
Enter the commands as follows:
(config)#voice trunk T01 type sip
(config-T01)#description PROVIDER
(config-T01)#sip-server primary 192.0.2.2
(config-T01)#trust-domain
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SBC and ShoreTel Interoperability
Configuring the ADTRAN SBC
Step 7: Configuring the ShoreTel PBX SIP Trunk
The second voice trunk that must be configured is the SIP trunk to the ShoreTel PBX from the ADTRAN
SBC. The trunk is configured using the voice trunk <Txx> type sip, description <text>, and
sip-server-primary <ipv4 address | hostname> commands. Use the sip-server-primary <ipv4 address |
hostname> command to set the server address to the ShoreTel PBX LAN1 IP address. In addition, the
ShoreTel PBX must control call transfers. This is accomplished using the transfer-mode-network
command in the trunk’s configuration. Use the grammar from host local command to specify that the IP
address of the interface is used in the SIP From header for outbound messages.
Enter the commands as follows:
(config)#voice trunk T11 type sip
(config-T11)#description SHORETEL
(config-T11)#sip-server primary 192.168.1.2
(config-T11)#hmr Shoretel_Outbound out
(config-T11)#trust-domain
(config-T11)#grammar from host local
(config-T11)#transfer-mode network
Step 8: Configuring a Trunk Group for the Service Provider
After configuring the two SIP trunks, configure an individual trunk group for the service provider trunk
account. The previously created trunks are added to the trunk group, which is then used to assign outbound
call destinations (local calls, long distance calls, etc.). A cost is also assigned to each accept template in
the trunk group.
Use the voice grouped-trunk <name> command to create a trunk group and to enter the Voice Trunk
Group Configuration mode. The trunk <Txx> command adds an existing trunk to the trunk group, so that
outbound calls can be placed out that particular trunk. The <Txx> parameter specifies the trunk identity
where xx is the trunk ID number.
Use the accept <pattern> command to specify number patterns that are accepted for routing calls out of
the trunk. Use the no form of this command to remove a configured dial pattern. The <pattern> parameter
is specified by entering a complete phone number or using wildcards to help define accepted numbers.
Valid characters for templates are as follows:
0-9
X
N
M
$
[]
,()
-
Match the exact digit(s) only
Match any single digit 0 through 9
Match any single digit 2 through 9
Match any single digit 1 through 8
Match any number string dialed
Match any digit in the list within the brackets (for example, [1,4,6])
Formatting characters that are ignored but allowed
Use within brackets to specify a range, otherwise ignored
The following are example template entries using wildcards:
1) NXX-XXXX
6AOSSG0009-42B
Match any 7-digit number beginning with 2 through 9
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SBC and ShoreTel Interoperability
Configuring the ADTRAN SBC
2) 1-NXX-NXX-XXXX
Match any number with a leading 1, then 2 through 9, then any 2 digits,
then 2 through 9, then any 6 digits
3) 555-XXXX
Match any 7-digit number beginning with 555
4) XXXX$
Match any number with at least 5 digits
5) [7,8]$
Match any number beginning with 7 or 8
6) 1234
Match exactly 1234
Some template number rules:
1. All brackets must be closed with no nesting of brackets and no wildcards within the brackets.
2. All brackets can hold digits and commas, for example: [1239]; [1,2,3,9]. Commas are implied between
numbers within brackets and are ignored.
3. Brackets can contain a range of numbers using a hyphen, for example: [1-39]; [1-3,9].
4. The $ wildcard is only allowed at the end of the template, for example: 91256$; XXXX$.
Enter the commands as follows:
(config)#voice grouped-trunk PROVIDER
(config-PROVIDER)#trunk T01
(config-PROVIDER)#accept NXX-NXX-XXXX cost 0
(config-PROVIDER)#accept 1-NXX-NXX-XXXX cost 0
(config-PROVIDER)#accept 011-$ cost 0
(config-PROVIDER)#accept 411 cost 0
(config-PROVIDER)#accept 611 cost 0
(config-PROVIDER)#accept 911 cost 0
Step 9: Configuring a Trunk Group for the ShoreTel PBX
After configuring a trunk group for the service provider, create a trunk group for the ShoreTel PBX trunk
account. Create the trunk group using the voice grouped-trunk <name> command. Add an existing trunk
to the trunk group using the trunk <Txx> command. The outbound allowed calls are defined using the
accept <pattern> command and are assigned a cost using the cost parameter, as described in
Step 8: Configuring a Trunk Group for the Service Provider on page 10.
Enter the commands as follows:
(config)#voice grouped-trunk SHORETEL
(config-SHORETEL)#trunk T11
(config-SHORETEL)#accept $ cost 0
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SBC and ShoreTel Interoperability
ADTRAN SBC Sample Configuration
ADTRAN SBC Sample Configuration
The following example configuration is for a typical installation of an ADTRAN SBC IP business gateway
or router with SIP trunking configured to the service provider and the ShoreTel PBX. This is the
configuration that was used to validate the interoperability between the ADTRAN SBC and the ShoreTel
PBX. Only the commands relevant to the interoperability configuration are shown.
The configuration parameters entered in this example are sample configurations only, and
only pertain to the configuration of the SIP trunking gateway functionality. This
application should be configured in a manner consistent with the needs of your particular
network. CLI prompts have been removed from the configuration example to provide a
method of copying and pasting configurations directly from this guide into the CLI. This
configuration should not be copied without first making the necessary adjustments to
ensure it will function properly in your network.
!
interface eth 0/1
description LAN
ip address 192.168.1.1 255.255.255.0
media-gateway ip primary
no shutdown
!
interface eth 0/2
description WAN
ip address 203.0.113.2 255.255.255.252
media-gateway ip primary
no shutdown
!
!
ip route 0.0.0.0 0.0.0.0 203.0.113.1
!
!
voice transfer-mode local
voice forward-mode local
!
hmr policy ShoreTel_Outbound
rule-set ShoreTel_Early_Media_Workaround 10
!
hmr rule-set ShoreTel_Early_Media_Workaround
message-rule Convert_183_to_180 message-type response 10
modify header sip-status-line position first match-value "/183 Session Progress/" new-value "/180
Ringing/" 10
!
voice trunk T01 type sip
description PROVIDER
sip-server primary 192.0.2.2
trust-domain
!
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SBC and ShoreTel Interoperability
Configuring the ShoreTel PBX
voice trunk T11 type sip
description SHORETEL
sip-server primary 192.168.1.2
hmr Shoretel_Outbound out
trust-domain
grammar from host local
transfer-mode network
!
voice grouped-trunk PROVIDER
trunk T01
accept NXX-NXX-XXXX cost 0
accept 1-NXX-NXX-XXXX cost 0
accept 011-$ cost 0
accept 411 cost 0
accept 611 cost 0
accept 911 cost 0
!
voice grouped-trunk SHORETEL
trunk T11
accept $ cost 0
!
!
ip rtp symmetric-filter
ip rtp media-anchoring
!
end
Configuring the ShoreTel PBX
The ShoreTel PBX is configured using the ShoreTel product’s GUI (ShoreWare Director). Refer to the
ShoreTel documentation for detailed instructions about accessing the GUI.
The following section describes ShoreTel system configuration to support SIP trunking interoperability
with the ADTRAN SBC. The section is divided into general system settings and trunk configurations (both
group and individual) needed to support SIP trunking.
The first configuration section pertains to general system settings and includes call control, site
administration, and switch administration. If these items have already been configured on the system, skip
this section and continue to Configuring System Settings on page 18.
ShoreTel points its individual SIP trunks to the ADTRAN SBC.
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SBC and ShoreTel Interoperability
Configuring the ShoreTel PBX
Configuring General Settings
To configure the General Settings for the ShoreTel PBX, follow these steps:
•
Configure Call Control on page 14
•
Configure Sites Administration on page 16
•
Configure Switch Administration on page 17
Configure Call Control
1. To configure the Call Control settings, log into ShoreWare Director and navigate to Administration >
Call Control > Options.
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SBC and ShoreTel Interoperability
Configuring the ShoreTel PBX
2. The Call Control Options menu will appear.
3. Ensure the Enable SIP Session Timer option is checked.
4. Set the Session Interval timer. The recommended session interval is 1800 seconds.
5. Select the appropriate refresher (from the drop-down menu) for the SIP Session Timer. Set the
Refresher field to either Caller (UAC) [User Agent Client] or Callee (UAS) [User Agent Server]. If
Caller (UAC) is selected, the caller’s device will be in control of the session timer refresh. If Callee
(UAS) is selected, the device of the person called will control the session timer refresh.
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SBC and ShoreTel Interoperability
Configuring the ShoreTel PBX
6. Uncheck the box for Always Use Port 5004 for RTP. For SIP configurations, Dynamic User Datagram
Protocol (UDP) must be used for RTP traffic. If the box is unchecked, Media Gateway Control Protocol
(MGCP) will no longer use UDP port 5004; MGCP and SIP traffic will use dynamic UDP ports.
The option Always Use Port 5004 for RTP will be grayed out by default if SIP servers,
SIP trunks, or SIP extensions are configured.
7. Reboot the following items: IP Phones, ShoreGear Switches, ShoreWare Director, Distributed Voice
Services/Remote Servers, Conference Bridges, and Contact Centers. If a full system reboot is not
performed, one-way audio will occur during initial testing.
Configure Sites Administration
1. To configure the sites settings (related to the administration of sites) navigate to Administration >
Sites.
2. In the Bandwidth section of the Sites Edit menu that appears, set the Admission Control Bandwidth.
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3. The Admission Control Bandwidth setting defines the bandwidth available to and from the site. This
is important because SIP devices will be counted against the site bandwidth. Bandwidth must be set
appropriately based on the site’s configuration with the service provider SIP trunking. Refer to the
ShoreTel Planning and Installation Guide for more information.
4. Set the Intra-Site Calls and the Inter-Site Calls settings next. For the Intra-Site Calls, verify that the
desired audio bandwidth is selected from the CODEC list for calls within the system. The settings
should then be confirmed for the desired audio bandwidth CODEC list for Inter-Site Calls (calls
between sites).
The CODEC list selection is used as an example. Refer to the ShoreTel Planning and
Installation Guide for additional information on CODEC list selections.
SIP uses both G.711 and G.729 CODECs. The CODEC setting will be negotiated to the
highest CODEC supported (fax requires G.711 at minimum).
Configure Switch Administration
1. The final general setting to configure is allocating ports for the switch settings. To make these changes,
navigate to Administration > Platform Hardware > Voice Switches / Service Appliances >
Primary.
2. From the Voice Switches menu that appears, select the name of the switch to configure. The Edit
ShoreGear Switch screen will display.
3. From the Edit ShoreGear Switch menu, select the desired number of SIP trunks from the ports
available.
4. Each port designated as a SIP trunk enables the support for five individual trunks.
5. Select the check box for Enable Jack Based Music On Hold if MOH is expected on the SIP trunk.
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For more information on configuring file based MOH, refer to the ShoreTel
Administration Guide available from the ShoreTel support website:
http://support.shoretel.com/.
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Configuring the ShoreTel PBX
Configuring System Settings
To configure the System Settings for the ShoreTel PBX, follow these steps:
•
Configure SIP Trunk Groups on page 19
•
Configure Trunk Services on page 22
•
Configure Individual Trunks on page 24
Configure SIP Trunk Groups
If the SIP trunk groups have already been configured on the system, proceed to the section Configure
Individual Trunks on page 24.
ShoreTel trunk groups support only static SIP endpoint individual trunks.
1. To modify the SIP trunk groups settings, navigate to Administration > Trunks > Trunk Groups.
2. From the drop-down menus on the Trunk Groups menu, select the desired site and select the SIP trunk
type to configure.
3. Select the Go link to the right of the Add new trunk group at site option. The Edit SIP Trunk Group
menu will appear.
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4. Select your service provider or Default ITSP profile from the drop-down list.
5. From the Edit SIP Trunks Group menu, enter a name for the trunk group. In the example below, the
name Provider has been created.
6. Ensure Enable SIP Info for G.711 DTMF Signaling is not checked (disabled).
The Digest Authentication field is not required when connecting to an ADTRAN SBC.
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7. From the Inbound settings at the bottom of the Edit SIP Trunks Group menu, ensure the Number of
Digits from CO is set to 10. Enable the DNIS or DID parameters as necessary for your installation. It
is not necessary to enable the Extension parameter for SIP Trunks, because it defaults to disabled, but
it can be enabled if desired (refer to the ShoreTel Planning and Installation Guide for further
information).
8. Enable Tandem Trunking if you plan on transferring calls to external parties via the SIP trunk.
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Configuring the ShoreTel PBX
Configure Trunk Services
1. From the Trunk Groups menu in the Trunk Services section, enable or disable the appropriate services
based on what the service provider supports and which features are required from this trunk group. The
parameter Enable Original Caller Information should be enabled.
2. From the Trunk Digit Manipulation section, make sure the Remove leading 1 from 1+10D parameter
is enabled (checked).
3. After these settings have been configured in the Edit SIP Trunk Group menu, select Save to accept
the changes.
4. This completes the settings necessary to configure the trunk groups on the ShoreTel system. Log out of
ShoreTel Director.
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5. You will then be presented with the ShoreTel Director login page. From this page, you will enable the
Support Entry mode of the ShoreTel Director. On your keyboard, hold down the <CTRL> and <Shift>
keys and with the mouse pointer click on the U of Username.
6. Log into ShoreTel Director with your usual administration user credentials.
7. Navigate to the Edit SIP Trunk Group menu, by selecting Administration > Trunks > Trunk
Groups.
8. From the Trunk Groups menu, select the trunk group you created for your service provider (in the
previous steps). The Edit SIP Trunk Group menu will appear.
9. Scroll down to the bottom of the menu to the Trunk Group Dialing Rules parameter section. Select
the Edit button to the right of the Custom parameter.
.
10. The Trunk Groups Dialing Rules – Webpage Dialog will appear. In the blank area of the webpage
dialog, enter, ;10E and select Save. Be sure to enter the exact syntax shown here. Include the
semicolon, one, and zero, followed by an uppercase E. This syntax is case sensitive, so verify that it
matches the screen below.
This completes the settings necessary to set up the trunk groups on the ShoreTel system.
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Configure Individual Trunks
1. To configure the individual trunks, navigate to Administration > Trunks > Individual Trunks.
2. From the Trunks by Group menu that appears, select the site for the new individual trunk(s) to be
added. Select the appropriate trunk group from the drop-down list in the Add new trunk at site area.
In this example, the site is Headquarters and the trunk group is SIP.
3.
Select the Go link to the right of the Add new trunk at site option to bring up the Edit Trunk screen.
4. From the individual trunks Edit Trunk menu, enter a name for the individual trunk. Select the
appropriate switch, SIP trunk type, and enter the number of trunks. When selecting a name, it is
recommended that you name the individual trunks the same as the name of the trunk group so that the
trunk type can be tracked easily.
5. Select the switch upon which the individual trunk will be created. For the service provider trunk, select
Use IP Address and enter the IP address of the ADTRAN SBC device.
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Troubleshooting
6. Select the number of individual trunks desired, each one supports one audio path. For example, if 5 is
entered, then 5 audio paths can be up at one time. Once these changes are complete, press Save to make
the changes.
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 ADTRAN SBC.
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. The ShoreTel PBX is now configured for
interoperability with the ADTRAN SBC gateway.
Troubleshooting
The ADTRAN SBC unit has several show and debug commands that display information on the console
session. The show commands display a snapshot of the activity on the ADTRAN SBC unit. The debug
commands display real time activity. Some useful show and debug commands are listed below.
Use the show ip rtp media sessions command to display all of the anchored RTP flow associations and the
number of relayed packets per association currently active in an anchored RTP flow. In addition, the and
the session type (digital signal processing (DSP), media-anchored, or transcoded) for the association is
displayed.
#show ip rtp media sessions
Use the debug voice switchboard command to display the decision making process as it routes incoming
and outgoing calls. The debug output of this command displays the trunk from which the call was received
and the trunk to which the call was routed.
#debug voice switchboard
Use the debug sip stack messages command to display the SIP messages that are received by or sent from
the ADTRAN SBC. This command displays the full SIP headers as well as the SDP body and will tell
whether the ADTRAN SBC received or transmitted the message.
#debug sip stack messages
Packet Capture
The Packet Capture capability of the ADTRAN SBC allows the capture and export of all traffic on any one
or all interfaces simultaneously. It can then be exported to your PC where it can be viewed in Wireshark or
Ethereal. For more information about configuring packet capture, refer to the configuration guide
Configuring Packet Capture in AOS.
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Additional Resources
Additional Resources
There are additional resources available to aid in configuring your ADTRAN SBC unit. Many of the topics
discussed in this guide are complex and require additional understanding, such as using the CLI, SBC in
AOS, and HMR. The documents listed in Table 7 are available online at ADTRAN’s Support Forum at
https://supportforums.adtran.com.
Table 7. Additional ADTRAN Documentation
Feature
Document Title
All AOS Commands Using the CLI
AOS Command Reference Guide
Sample Configuration for SIP Trunking on
ADTRAN SBC
SBC SIP Trunking Sample Configuration
SBC Product Overview
Session Border Controllers in AOS
Media Anchoring
Configuring Media Anchoring in AOS
Firewall (IPv4)
Configuring the Firewall (IPv4) in AOS
Security Settings
Security Best Practices for AOS Products
Access Control Lists
Configuring IP Access Control Lists (ACLs) in AOS
Access Control Policies
Configuring Access Policies in AOS
HMR
Manipulating SIP Headers and Messages in AOS
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