This document and its contents was prepared by:
SHORETEL APPLICATION NOTE
for
AT&T IP Flexible Reach SIP Trunking via Ingate
Date: July 7, 2016
App Note Number: TC-16049
For use with: AT&T IP Flexible Reach SIP Trunking via Ingate
Product: ShoreTel Connect ONSITE
System: ShoreTel Connect ONSITE
Build 21.77.3614.0
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Contents
Audience ......................................................................................................................... 3
SIP Trunking Network Components ................................................................................ 4
Features .......................................................................................................................... 6
Configuration ................................................................................................................... 7
ShoreTel - Create Custom Codec Lists and Sites ................................................ 8
ShoreTel - SIP Trunk Configuration ................................................................... 10
Ingate – Configuration SIP Trunking via Tool TG ............................................... 15
Ingate – Apply and Edit SIP Trunking Configuration via Web GUI ..................... 22
Summary of Tests and Results ..................................................................................... 32
Conclusion .................................................................................................................... 41
Additional Resources .................................................................................................... 41
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Audience This document is intended for the SIP Trunk Customer’s technical staff and Value Added Reseller
(VAR) having installation and operational responsibilities
Introduction
This Application Note describes the configuration steps for AT&T IP Flexible Reach SIP Trunking
with ShoreTel Connect ONSITE System via Ingate SIParator/Firewall
AT&T
Service Providers today, such as AT&T, are offering alternative methods to connect to the PSTN
via their IP network. Most of these services utilize SIP as the primary signaling method and a
centralized IP to TDM gateway to provide on-net and off-net services. AT&T is a service provider
offering that allows connection to the PSTN and may offer the end customer a viable alternative to
traditional PSTN connectivity via either analog or T1 lines.
To contact AT&T sales or support, please visit
http://www.att.com/support
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SIP Trunking Network Components
The network for the SIP Trunk reference configuration is illustrated below and is representative of a
ShoreTel Connect ONSITE System configuration
10.89.9.0/24
AT&T NetworkCustomer Premise
SIP SIP
ShoreTel Voice Switch ST100A
10.89.9.5
ShoreTel Server
10.89.9.2
Fax/Analog
ShoreTel Virtual Trunk Switch
10.89.9.6
ShoreTel Virtual Phone Switch
10.89.9.4
ShoreTel IP480
ShoreGear 9010.89.9.7
Fax/Analog
ShoreTel IP560
PSTN
Ingate SIParatorLAN: 10.65.1.100
ShoreTel Virtual Appliance
Collaboration10.89.9.8
ShoreTel Connect Client
Bria 4 Desktop Client
ShoreTel Connect Contact Center
10.89.9.3
AT&T Network
Figure 1: SIP Trunk Lab Reference Network
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Hardware Components
VMware ESXi 5.5
ShoreTel Voice Switch (ST100DA)
ShoreTel ShoreGear Switch (SG90)
Analog Fax Machine
ShoreTel 560 IP Phones
ShoreTel 480/485G IP Phones
ShoreTel Collaboration Service Appliance
ST Connect Contact Center
Software Requirements
ShoreTel Connect ONSITE Build 21.77.3614.0
ShoreTel Connect Contact Center Build 507.80.6109.0
ShoreTel Connect Client 212.4000.3559.0
Windows Server 2012 R2 Standard
Ingate SIParator/Firewall 5.0.9
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Features Features Supported
Basic calls with G711Ulaw
Call Hold and Resume
Call Transfer (with re-invite)
Call Forwarding
Make Me Conference
DTMF RFC 2833
Calling Party Number Presentation
Calling Party Number Restricted
Hunt Group
Group Pickup
Call Park/un-park
Simul Ring
Call Forward – “FindMe”
Call Recording
Auto-Attendant
Work Group
Office Anywhere External
Silent Monitor / Barge-In / Whisper Page
Contact Center
Conference bridge
Fax G711 Pass-through
Features Not Supported
SIP REFER Support: AT&T requires blind and consultative transfers using REFER. Call
transfers using SIP REFER are not supported.
Caveats and Limitations
T.38 fax is not supported with the new ShoreTel Voice Switches (ShoreTel ST-100DA), and
with ShoreTel Virtual Trunk Switches. T.38 fax is supported with ShoreTel Legacy
Hardware Switches (SG-90). Known issue ENG-189880.
NOTE: There may be other feature limitations when using SIP Trunks. Please refer to Chapter 19 of
the ShoreTel Connect ONSITE Administration Guide for more information.
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Configuration
Configuration Steps
In this section an overview is presented of the steps that are required to configure the ShoreTel
Connect ONSITE IP-PBX and Ingate SIParator for SIP Trunking to AT&T.
Table 1 – PBX Configuration Steps
Step Description
Step 1 Codec Lists and Sites
Step 2 SIP Trunk Configuration
Table 2 – Ingate SIParator Configuration Steps
Step Description
Step 1 SIP Trunking via Tool TG Configuration
Step 2 Apply and Edit SIP Trunking Configuration via Web GUI
IP Address Worksheet
The specific values listed in the table below and in subsequent sections are used in the lab configuration described in this
document, and are for illustrative purposes only. The customer must obtain and use the values for your deployment.
Table 3 – IP Address Worksheet
Component MSO Lab Value Customer Value
ShoreTel Connect ONSITE IP-PBX
ShoreTel Server 10.89.9.2 Unique to every deployment
ShoreTel Voice Switch ST100DA 10.89.9.5 Unique to every deployment
ShoreTel Virtual Trunk Switch 10.89.9.6 Unique to every deployment
ShoreTel ShoreGear 90 10.89.9.7 Unique to every deployment
ShoreTel Virtual Phone Switch 10.89.9.4 Unique to every deployment
ShoreTel Virtual Contact Center 10.89.9.3 Unique to every deployment
ShoreTel Virtual Appliance Collaboration
10.89.9.8 Unique to every deployment
Ingate SIParator
LAN IP Address 10.65.1.100 Unique to every deployment
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ShoreTel - Create Custom Codec Lists and Sites
Create Codec Lists
1. Navigate to Features > Call Control > Codec Lists
2. Click NEW
3. Set Description: AT&T is used for this example
4. Codec List Members: G729/8000 and PCMU/8000 were chosen and added for this
test
5. Click SAVE
Figure 2: Codec Lists
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Create Sites
1. Navigate to System > Sites
2. Set Name: Headquarters
3. Set Local Area Code: 214 is used in this test
4. Set Admission Control Bandwidth: 2048
5. Set Intra-Site Calls: Newly created Codec List, AT&T, was selected from the drop down
menu as an example
6. Set Inter-Site Calls: Codec List, AT&T, is selected from the drop down menu
7. Set Fax and Modem Calls: Default Codec List, Fax Codecs – Low Bandwidth Pass-
through is selected from the drop down menu
8. Leave all other fields as default
9. Click SAVE
Figure 3: Sites
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ShoreTel - SIP Trunk Configuration
This section describes the ShoreTel configuration necessary to support connectivity to AT&T SIP
Trunking service
Create SIP Trunk Profiles SIP Profile is critical for SIP Trunking deployment. The Default ITSP SIP Profile is used for this
test.
1. Navigate to Trunks > SIP Profiles
2. Check Default ITSP under NAME
Figure 4: SIP Trunk Profiles
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Add Trunk Group
1. Navigate to Trunks > Trunk Groups > Trunk Groups
2. Select the GENERAL tab
3. Set Name: ATT
4. Set Trunk Type: SIP
5. Set Profile: Default ITSP SIP Profile is selected from drop down menu
6. Click SAVE
Figure 5: Trunk Groups
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7. Select the INBOUND tab
8. Set Number of Digits from CO: 10 is used in this setup
9. DNIS: Checked
10. DID: Checked
Figure 6: Trunk Groups – Cont.
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11. Select the OUTBOUND tab
12. Outgoing: Checked
13. Set Access Code: 9 is used in this example
14. Set Local Area Code: 214 is used in this example
15. Set Billing Telephone Number: Pilot number will be provided by your AT&T Account
Representative and must be kept confidential
16. Caller ID Not Blocked by Default: Checked
17. Dial 7 digits for Local Area Code: Unchecked
18. Leave all other fields as default
19. Click SAVE
Figure 7: Trunk Groups – Cont.
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Create Individual Trunks
1. Navigate to Trunks > Trunks
2. Click NEW
3. Set Site: Headquarters
4. Set Trunk Group: ATT (SIP) is selected from the drop down menu
5. Set Name: AT&T is used in this setup
6. Set Switch: Lab109-vTS1 is selected from the drop down menu
7. Set IP Address or FQDN: Enter the LAN IP Address of Ingate SIParator. Please contact
your AT&T or ShoreTel Business Representative for more information regarding your
deployment.
8. Set Number of Trunks: 4 is given in this setup
9. Click SAVE
Figure 8: Individual Trunks
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DTMF Payload configuration
1. Navigate to Features > Call Control > Options
2. Under the General options, scroll down to the DTMF/RFC-2833 payload type
3. Set 101 as the payload type
4. Save the configuration
Figure-DTMF
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Ingate – Configuration SIP Trunking via Tool TG
1. Launch Ingate SUT (Startup Tool) TG
2. Please Select Model: Ingate Firewall/SIParator is selected from the drop down
3. Click Next
Figure 9: Product Type Selection
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4. Configure SIP Trunking: Checked
5. Set Password: Enter the proper Ingate password provided for your deployment
6. Click Contact
Figure 10: Ingate Startup Tool
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7. Select the Network Topology tab
8. Verify the network parameters are properly set
Figure 11: Network Topology
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9. Select the IP-PBX tab
10. Set Type: ShoreTel ShoreGear is selected from drop down
11. Set IP Address: Enter the IP Address of the ShoreTel Connect SIP Trunk Switch
Figure 12: IP-PBX
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12. Select the ITSP_1 tab
13. Set Name: Generic(no register) is selected from drop down menu for this test
14. Set IP Address: Please contact your AT&T for your deployment
Figure 13: ITSP
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15. Select the Upload Configuration tab
16. Logon to web GUI and apply settings: Checked
17. Click Upload
Figure 14: Upload Configuration
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Ingate – Apply and Edit SIP Trunking Configuration via Web GUI
1. After the configuration is uploaded via the SUT TG, login to Ingate SIParator/Firewall via
web GUI
2. Set Username and Password: Enter proper username and password for Ingate
Figure 15: Login to Ingate SIParator via Web GUI
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3. Navigate to Administration > Save/Load Configuration
4. Click Apply Configuration to apply the changes
Figure 16: Apply Configuration
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5. Click Save Configuration to save the change to the SIParator/Firewall
Figure 17: Save Configuration
6. Select the SIP Trunks tab
7. Set View Trunk: Select the relevant SIP trunk from the drop down menu.
SIP Trunk 1: Generic (no register); ShoreTel ShoreGear is selected for this setup
8. Click Goto SIP Trunk Page
Figure 18: SIP Trunks
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9. Most of the parameters in SIP Trunk are defined through the SUT TG. Verify the
parameters and make changes accordingly.
Figure 19: SIP Trunks – Cont.
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10. Ensure Relay media for remote users is set to No. If it is set to YES, one-way audio will
occur when the ShoreTel phone transfers the incoming call to another PSTN user while
utilizing ShoreGear 90.
Figure 20: SIP Trunks – Cont.
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1. Under Main Trunk Line Section, update the Forward to field for the incoming value to the
following value:
$1?From=%3csip%3a$(from.user)%4010.65.1.100%3e%3buser%3dphone&P-Asserted-
Identity=%3csip%3a$(from.user)%4010.65.1.100%3e%3buser%3dphone&Contact=%3csip
%3a$(from.user)%4010.65.1.100%3e%3buser%3dphone&!Contact=%3csip%3a$(to.user)
%4010.65.1.100%3e%3buser%3dphone
2. Under PBX Lines Section, update the User Name field for Outgoing Calls to the following
value for the non-anonymous row: [email protected]? P-Asserted-
Identity=%3csip%3a$(from.user)%40192.65.xx.xx%3e%3buser%3dphone&Contact=%3csi
p%3a$(from.user)%40192.65.xx.xx%3e%3buser%3dphone
Figure 21: SIP Trunks – Cont.
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3. All other parameters are set to default values
4. Click Save
Figure 22: SIP Trunks – Cont.
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5. Navigate to SIP Services > Sessions and Media
6. Under Session Configuration, set Session Timer: 900 seconds is given for this test
Figure 23: Sessions & Media
7. Under Requests, set Default timeout for INVITE requests: 300 seconds is given for this
test. If this was set too low, Ingate will disconnect the call when the timer expires and the
called party has not answered the call
8. Click Save
Figure 24: Sessions & Media – Cont.
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Ingate responds to SIP OPTION by default. In order to pass the SIP OPTION from ShoreTel to
AT&T, complete the following configuration:
9. Navigate to SIP Traffic > Dial Plan
10. Under the Matching Request-URI section, click Add new rows
11. Set Name: AT&T-OPT is given in this setup
12. Set Tail: nothing is selected from drop down
13. Set Domain: Enter the Ingate LAN IP address assigned for your deployment
Figure 25: Matching Request-URI
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14. Under the Forward To section, click Add new rows
15. Set Name: AT&T-OPT is given in this setup
16. Set Replacement Domain
17. Set Port: 5060 which is the SIP Port used in this setup
18. Set Transport: UDP is given for this test
19. Under the Dial Plan section, click Add new rows
20. Set From Header: Select OPT from the drop down
21. Set Request-URI: Select newly configured Matching Request-URI AT&T-OPT from the
drop down
22. Set Action: Select Forward from the drop down
23. Set Forward To: Newly configured AT&T-OPT
24. Click Save at the bottom of the page
Figure 26: Forward SIP OPTION
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Summary of Tests and Results N/S = Not Supported N/T= Not Tested N/A= Not Applicable C.PASS= Conditional Pass
Primary Switch Test Plan (ST Virtual Trunk Switch)
ID Result Name Description Notes
1.1 PASS Setup and
Initialization
Verify successful setup and
initialization of the SUT
1.2 PASS Outbound Call
(Domestic)
Verify calls outbound placed through
the SUT reach the external
destination
1.3 PASS Inbound Call
(Domestic)
Verify calls received by the SUT are
routed to the default trunk group
destination
1.4 PASS All Trunks
Busy– Inbound
Caller
Verify an inbound caller hears busy
tone when all channels/trunks are in
use
1.5 PASS All Trunks
Busy–
Outbound
Caller
Verify an outbound caller hears busy
tone when all channels/trunks are in
use
1.6 PASS Incomplete
Inbound Calls
Verify proper call progress tones are
provided and proper call teardown for
incomplete inbound calls
ID Result Name Description Notes
2.1 C.PASS Codec
Negotiation
Verify codec negotiation between the
SUT and the calling device with each
side configured for a different codec
ShoreTel defect
ENG-349616
2.2 PASS 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
2.3 PASS Auto Attendant
Menu
Verify that inbound calls are properly
terminated on the ShoreTel Auto
Attendant menu and that you can
transfer to the desired extension
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ID Result Name Description Notes
2.4 PASS Auto Attendant
Menu checking
Voicemail
mailbox
Verify that inbound calls are properly
terminated on the ShoreTel Auto
Attendant menu and that you can
transfer to the Voicemail Login
Extension
ID Result Name Description Notes
3.1 PASS Post Dial Delay Verify that post dial delay is within
acceptable limits
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ID Result Name Description Notes
4.1 PASS Caller ID Name
and Number -
Inbound
Verify that Caller ID name and
number is received from SIP
endpoint device
4.2 PASS Caller ID Name
and Number -
Outbound
Verify that Caller ID name and
number is sent from SIP endpoint
device
4.3 PASS Hold from SUT
to SIP
Reference
Verify successful hold and resume
of connected call
4.4 PASS Call Forward -
SUT
Verify outbound calls that are being
forwarded by the SUT are redirected
and connected to the appropriate
destination
4.5 PASS Call Forward –
External PSTN
Number
Verify outbound calls that are being
forwarded by the SUT are redirected
and connected to the appropriate
destination
4.6 PASS Call Transfer –
Blind
Verify a call connected from the
SUT to the ShoreTel phone can be
transferred to an alternate
destination
4.7 PASS Call Transfer –
Consultative
Verify a call connected from the
SUT to the ShoreTel phone can be
transferred to an alternate
destination
4.8 PASS Outbound 911 Verify that outbound calls to 911 are
routed to the correct PSAP for the
calling location and that caller ID
information is delivered
4.9 PASS Operator
Assisted
Verify that 0+ calls are routed to an
operator for calling assistance
4.10 PASS 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
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ID Result Name Description Notes
4.11 PASS 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
4.12 PASS 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
4.13 PASS Inbound call to
DNIS/DID and
leave a voice
mail 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
4.14 PASS Call Forward –
“FindMe”
Verify that inbound calls are
forwarded to a user’s “FindMe”
destination
4.15 N/T Inbound /
Outbound Fax
Calls
Verify that inbound / outbound fax
calls complete successfully
4.17 PASS Inbound call to
a Group Pickup
Extension
Verify that inbound calls properly
presented to all of the phones that
have Group Pickup configured and
that the call can be answered,
placed on-hold and then transferred
4.18 PASS Office
Anywhere
External
Verify that inbound calls are
properly presented to the Office
Anywhere External PSTN
destination
4.19 PASS Simul Ring Verify that inbound calls are
properly presented to the desired
extension and the “Additional
Phones” destinations
4.20 PASS MakeMe
Conference
Verify that an inbound call can be
conferenced with three (or more)
additional parties
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ID Result Name Description Notes
4.21 PASS Park / Unpark Verify that an inbound call can be
parked and unparked
4.22 PASS Call Recording Verify that external calls can be
recorded via the SIP Trunk using
ShoreTel Communicator
4.23 PASS Silent Monitor /
Barge-In /
Whisper Page
Verify that external calls can be
silently monitored, barged-in and
whisper paged via the SUT
4.24 PASS Long Duration –
Inbound
Verify that an inbound call is
established for a minimum of 30
minutes
4.25 PASS Long Duration –
Outbound
Verify that an outbound call is
established for a minimum of 30
minutes
ID Result Name Description Notes
5.1 N/A Registration or
Digest
Authentication
Verify the SUT supports the use of
registration or digest authentication
for service access for inbound and
outbound calls
This testing didn’t
require registration/
authentication
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Secondary Switch Sanity Test Results (ST Voice Switch)
ID Result Name Description Notes
1.2 PASS Outbound Call
(Domestic)
Verify calls outbound placed through
the SUT reach the external
destination
1.3 PASS Inbound Call
(Domestic)
Verify calls received by the SUT are
routed to the default trunk group
destination
2.2 PASS DTMF
Transmission –
Out of Band /
In Band
Verify transmission of in-band and
out-of-band digits per RFC 2833 for
various devices connected to the
SUT
4.5 PASS Call Forward –
External PSTN
Number
Verify outbound calls that are being
forwarded by the SUT are redirected
and connected to the appropriate
destination
4.6 PASS Call Transfer –
Blind
Verify a call connected from the
SUT to the ShoreTel phone can be
transferred to an alternate
destination
4.7 PASS Call Transfer –
Consultative
Verify a call connected from the
SUT to the ShoreTel phone can be
transferred to an alternate
destination
4.12 PASS 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
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4.15 Failed Inbound /
Outbound Fax
Calls
Verify that inbound / outbound fax
calls complete successfully
Inbound T.38 fax call
fails, ShoreTel sends
488 not acceptable
here for the T.38 Re-
Invite from PSTN
4.16 PASS ShoreTel
Service
Appliance
Unified
Communication
System
Verify that inbound calls are
properly forwarded to the ShoreTel
Service Appliance and it properly
accepts the access code and you’re
able to participate in the conference
bridge
4.21 PASS MakeMe
Conference
Verify that an inbound call can be
conferenced with three (or more)
additional parties
4.23 PASS Call Recording Verify that external calls can be
recorded via the SIP Trunk using
ShoreTel Communicator
4.24 PASS Silent Monitor /
Barge-In /
Whisper Page
Verify that external calls can be
silently monitored, barged-in and
whisper paged via the SUT
4.27 PASS Contact Center Verify that an inbound call can be
established directly to the ShoreTel
Contact Center, that all prompts are
heard and the agent can answer the
call
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Tertiary Switch Sanity Test Results (ST Legacy Switch)
ID Result Name Description Notes
1.2 PASS Outbound Call
(Domestic)
Verify calls outbound placed
through the SUT reach the
external destination
1.3 PASS Inbound Call
(Domestic)
Verify calls received by the SUT
are routed to the default trunk
group destination
2.2 PASS DTMF
Transmission –
Out of Band /
In Band
Verify transmission of in-band
and out-of-band digits per RFC
2833 for various devices
connected to the SUT
4.5 PASS Call Forward –
External PSTN
Number
Verify outbound calls that are
being forwarded by the SUT are
redirected and connected to the
appropriate destination
4.6 PASS Call Transfer –
Blind
Verify a call connected from the
SUT to the ShoreTel phone can
be transferred to an alternate
destination
4.7 PASS Call Transfer –
Consultative
Verify a call connected from the
SUT to the ShoreTel phone can
be transferred to an alternate
destination
4.12 PASS 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
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4.15 PASS Inbound /
Outbound Fax
Calls
Verify that inbound / outbound fax
calls complete successfully
4.21 PASS MakeMe
Conference
Verify that an inbound call can be
conferenced with three (or more)
additional parties
4.23 PASS Call Recording Verify that external calls can be
recorded via the SIP Trunk using
ShoreTel Communicator
4.24 PASS Silent Monitor /
Barge-In / Whisper
Page
Verify that external calls can be
silently monitored, barged-in and
whisper paged via the SUT
4.27 PASS Contact Center Verify that an inbound call can be
established directly to the
ShoreTel Contact Center, that all
prompts are heard and the agent
can answer the call
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Conclusion
AT&T IP Flexible Reach SIP Trunking has been successfully tested with ShoreTel Connect
ONSITE via Ingate SIParator/Firewall Build 21.77.3614.0
Additional Resources ShoreTel Connect ONSITE Administration Guide
Ingate Administration Guide
Version Date Contributor Content
1.0 June 20, 2016 Satheesh Kumar Original Release
World Headquarters +1 (800) 425-9385 Toll Free EMEA APAC
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Sunnyvale, CA 94085 +1 (408) 331-3333 Fax Easthampstead Road Suntec Tower 3
USA Bracknell, RG12 1YQ Singapore 038988
shoretel.com +44 (0) 1344 208800 Tel +65 6517 0800 Tel
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