Geos Communications SIP Trunking

Transcription

T P P
A P P
N O T E
TPP-10253
Date: January, 2010
Product: ShoreTel | Ingate | Geos
Communications
System version: ShoreTel 9.2
Application Note
Geos Communications
23 February 2010
960 Stewart Drive Sunnyvale, CA 94085 USA Phone +1.408.331.3300 +1.877.80SHORE Fax +1.408.331.3333 www.ShoreTel.com
Table of Contents
APPLICATION NOTE .......................................................................................................................... 1
GEOS COMMUNICATIONS ............................................................................................................... 1
1
OVERVIEW .................................................................................................................................. 3
1.1
2
GEOS COMMUNICATIONS ..................................................................................................... 5
2.1
2.2
2.3
3
GEOS COMMUNICATIONS PRODUCT INFORMATION..................................................................... 5
GEOS COMMUNICATIONS CONFIGURATION ................................................................................ 5
GEOS COMMUNICATIONS SALES & SUPPORT .............................................................................. 5
INTEROPERABILITY REQUIREMENTS, CERTIFICATION AND LIMITATIONS ...... 6
3.1
3.2
3.3
4
SIP TRUNKING ARCHITECTURE OVERVIEW ................................................................................ 4
VERSION SUPPORT ...................................................................................................................... 6
CERTIFICATION RESULTS SUMMARY .......................................................................................... 7
LIMITATIONS AND OBSERVATIONS ........................................................................................... 10
SHORETEL ................................................................................................................................ 11
4.1 SHORETEL UNSUPPORTED FEATURES ....................................................................................... 11
4.2 SHORETEL CONFIGURATION ..................................................................................................... 12
4.2.1
ShoreTel System Settings – General: .............................................................................. 12
5
INGATE ....................................................................................................................................... 22
5.1 INGATE PRODUCT INFORMATION .............................................................................................. 22
5.2 INGATE PRODUCT CONFIGURATION .......................................................................................... 23
5.2.1
Startup Tool .................................................................................................................... 23
5.3 INGATE TROUBLESHOOTING ..................................................................................................... 30
5.3.1
Call Flow Examples ........................................................................................................ 30
5.3.2
Startup Tool .................................................................................................................... 31
5.3.3
Ingate Example Configuration........................................................................................ 35
5.3.4
Ingate Troubleshooting Tools ......................................................................................... 39
5.4 INGATE SALES & TECHNICAL SUPPORT .................................................................................... 42
5.4.1
Sales................................................................................................................................ 42
5.4.2
Technical Support ........................................................................................................... 42
6
DOCUMENT AND SOFTWARE COPYRIGHTS .................................................................. 43
7
TRADEMARKS.......................................................................................................................... 43
8
DISCLAIMER ............................................................................................................................ 43
9
COMPANY INFORMATION ................................................................................................... 43
pg. 2
1 Overview
This document provides details for connecting the ShoreTel system though the Ingate
SIParator® to Geos Communications for SIP Trunking to enable audio
communications. The document specifically focuses on the configuration procedures
needed to set up these systems to interoperate.
SIP Trunking allows the use of Session Initiation Protocol (SIP) communications
from an Internet Telephony Service Provider (ITSP) instead of the typical analog,
Basic Rate Interface (BRI), T1 or E1 trunk connections. Having the pure IP trunk to
the ITSP 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 Ingate SIParator which is connected to both the LAN and WAN and acts
as a secure gateway to Geos Communications for SIP Trunking.
ShoreTel and Ingate have teamed up to build a solid security focused solution,
ShoreTel being the IP PBX which sits on the LAN and connects to the Ingate
SIParator® / Firewall. Providing a solution to allow customers the ability to connect
to SIP Trunks offered by Geos Communications in a secure manner is important. The
Ingate 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:
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
pg. 3
1.1 SIP Trunking Architecture Overview
There are a number of different network deployments of SIP Trunking from a Service
Provider to an Enterprise. Some Service Providers provide SIP Trunking directly
over the Internet and others over provide managed link into their network.
Here are two typical deployments examples:
pg. 4
2 Geos Communications
GEOS Communications, Inc. is a leading provider of award-winning patented and
innovative high-quality mobile applications and services that employ best-of-breed
VoIP technology and use a combination of the Company's own services network and
the Internet to deliver high-quality communications products to customers on a global
scale. GEOS Communications’ current solutions, which use the MyGlobalTalk™
brand name, provide reduced-cost international long-distance calling from any device:
smartphone, cell phone, landline or desk phone, and/or computer.
2.1 Geos Communications Product Information
GEOS Communications’ SIP Trunking solution employs the Session Initiation
Protocol (SIP) in an interconnected deployment between an Enterprise’s IP-PBX and
the GEOS network. SIP Trunking, through the deployment of SIP-based
communications, permits the customer to utilize hi-reliability and lower cost IP
communications channels (“trunks) in-lieu of the typically higher cost analog Basic
Rate Interface (BRI), T1, or E1 lines when interconnected to the Public Switched
Telephone Network (PSTN).
2.2 Geos Communications Configuration
GEOS Communications supports common SIP Trunking configurations (see figure
below). In addition to interfacing an Enterprise’s IP-PBX to its network GEOS
employs the use of the InGate SIParator. With the SIParator deployed at the customer
network edge, GEOS is able to interface with any SIP-enabled IP-PBX; thereby
eliminating the need to achieve interoperability with every IP-PBX vendor and
address the widest number of customer environments. In this context, the SIParator
serves as a “normalization engine” between the IP-PBX and GEOS, ensuring a
successful SIP trunk roll-out.
2.3 Geos Communications Sales & Support
For GEOS Communications SIP Trunking Solutions Sales & Support please contact
the following:
Sales: Mr. David Schafer; 949.300.6322; [email protected]
Support: Mr. Steven Short; 678-608-3969; [email protected]
pg. 5
3 Interoperability Requirements, Certification
and Limitations
3.1 Version Support
Products are certified via the Technology Partner Certification Process for the
ShoreTel system. The table below contains the matrix of Ingate Firewall and Ingate
SIParator and GEOS Communications versions firmware releases certified on the
identified ShoreTel software releases.
Ingate Firewall and Ingate SIParator version
4.5.1 with
the patch igpatch-4-5-1shoretel-2
applied
4.5.2
4.6.1
4.7.1
4.8.1
ShoreTel 7.0





ShoreTel 7.5





ShoreTel 8.0





ShoreTel 8.1





ShoreTel 9.0





ShoreTel 9.2





Network gear used in the ITSP’s network along with the applicable releases which
were tested.
GEOS Communications Network Gear & Versioning
FreeSwitch
ver. 1.0.3
FreeSwitch
ver. 1.0.4
Cisco
7200vxr
ShoreTel 7.0



ShoreTel 7.5



ShoreTel 8.0



ShoreTel 8.1



ShoreTel 9.0



ShoreTel 9.2



pg. 6
3.2 Certification Results Summary
Here are TTP Test Plan Results when testing with <Insert ITSP Name Here>
Table 1: Initialization and Basic Calls
ID
Name
Description
1.1
Setup and
initialization
Outbound Call
(Domestic)
Inbound Call
(Domestic)
Verify successful setup and initialization
of the system under test (SUT)
Verify calls outbound placed through
the SUT reach the external destination.
Verify calls received by the SUT are
routed to the default trunk group
destination.
Verify that the SUT recovers after
power loss to the SUT
Verify the SUT recovers after loss of
network link to the SUT.
Verify an inbound caller hears busy tone
when all channels/trunks are in use
Verify an outbound caller hears busy
tone when all channels/trunks are in use
Verify proper call progress tones are
provided and proper call teardown for
incomplete inbound calls.
1.2
1.3
1.4
1.5
1.6
1.7
1.8
Device Restart –
Power Loss
Device Restart –
Network Loss
All Trunks Busy –
Inbound Callers
All Trunks Busy –
Outbound Callers
Incomplete
Inbound Calls
Notes
PASS
PASS
PASS
PASS
PASS
PASS
PASS
PASS
Table 2: Media and Dual-Tone Multi-Frequency (DTMF) Support
ID
Name
2.1
Error! Reference Error! Reference source not found.
source not found.
2.2
Media Support –
SIP Reference to
SUT
2.3
Codec Negotiation
2.4
DTMF
Transmission – Out
of Band / In Band
Auto Attendant
Menu
2.5
2.6
Auto Attendant
Menu “Dial by
Name”
Description
Verify call connection and audio path from
SIP Reference phones to an external
destination through the service provider
using all supported codes with both sides set
to a common codec.
Verify codec negotiation between the SUT
and the calling device with each side
configured for a different codec.
Verify transmission of in-band and out-ofband digits per RFC 2833 for various
devices connected to the SUT.
Verify that inbound calls are properly
terminated on the ShoreTel Auto Attendant
menu and that you can transfer to the
desired extension.
menu and that you can transfer to the
desired extension using the “Dial by Name”
feature.
Notes
PASS
PASS
PASS
PASS
PASS
PASS
pg. 7
ID
Name
Description
2.7
Auto Attendant
Menu Checking
Voice Mail Mailbox
menu and that you can transfer to the Voice
Mail Login Extension.
Notes
PASS
Table 3: Performance & Quality of Service
ID
Name
3.1
Voice Quality
Service Levels
3.2
Capacity Test
3.3
Post Dial Delay
3.4
Billing Accuracy
Description
Notes
Verify the SUT can provide a voice
quality service-level agreement (SLA)
across the WAN from the customer
premises to the SUT SIP gateway.
Verify the service provider interface can
sustain services through period of heavy
outbound and inbound load.
Verify that post dial delay is within
acceptable limits.
Verify that all test calls made are
accurately reflected in the SUT’s Call
Detail Record (CDR) and billing
reports.
N/T
PASS
PASS
PASS
Table 4: Enhanced Services and Features
ID
Name
Description
4.1
Caller ID Name
and Number Inbound
Caller ID Name
and Number Outbound
Hold from SUT to
SIP Reference
Call Forward SUT
Verify that Caller ID name and number
is received from SIP endpoint device
4.2
4.3
4.4
4.5
Call Transfer –
Blind
4.6
Call Transfer –
Consultative
4.7
Conference – Ad
Hoc
Notes
PASS
Verify that Caller ID name and number
is sent from SIP endpoint device
PASS
Verify successful hold and resume of
connected call
Verify outbound calls that are being
forwarded by the SUT are redirected
and connected to the appropriate
destination.
Verify a call connected from the SUT to
the ShoreTel phone can be transferred to
an alternate destination.
Verify a call connected from the SUT to
the ShoreTel phone can be transferred to
an alternate destination.
Verify successful ad hoc conference of
three parties
PASS
PASS
PASS
PASS
PASS
pg. 8
ID
Name
Description
4.8
Verify the SUT provides inbound
“dialed number information” and is
correctly routed to the configured
destination.
4.9
Inbound Direct
Inward
Dialing/Dialed
Number
Identification
Service
(DID/DNIS)
Outbound 911
4.10
Operator Assisted
4.11
Inbound /
Outbound call
with Blocked
Caller ID
Inbound call to a
Hunt Group
4.12
4.13
Inbound Call to a
Workgroup
4.14
Inbound Call to
DNIS/DID and
Leave a Voice
Mail Message
Call Forward –
“FindMe”
Call Forward
Always
4.15
4.16
4.17
4.18
Inbound /
Outbound Fax
Calls
ShoreTel
Converged
Conferencing
Server
Notes
PASS
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.
Verify that 0+ calls are routed to an
operator for calling assistance.
Verify that calls with Blocked Caller ID
route properly and the answering phone
does not display any Caller ID
information.
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.
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.
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.
Verify that inbound calls are forwarded
to a user’s “FindMe” destination.
Verify that inbound calls are
immediately automatically forwarded to
a user’s external destination.
Verify that inbound/outbound fax calls
complete successfully.
N/T
forwarded to the ShoreTel Converged
Conferencing Server, that it properly
accepts the access code, and you’re able
to participate in the conference bridge.
N/T
N/T
PASS
PASS
PASS
PASS
PASS
PASS
N/T
pg. 9
ID
Name
Description
4.19
Inbound Call to
Bridged Call
Appearance
(BCA) Extension
4.20
Inbound Call to a
Group Pickup
Extension
Verify that inbound calls properly
presented to all of the phones that have
BCA configured and that the call can be
answered, placed on-hold and then
transferred.
presented to all of the phones that have
Group Pickup configured and that the
call can be answered, placed on-hold
and then transferred.
Notes
PASS
PASS
Table 5: Security
ID
Name
Description
5.1
Digest
Authentication
Verify the SUT supports the use of
digest authentication for service access
for inbound and outbound calls.
Notes
N/T
3.3 Limitations and Observations
Observations in Scale-ability
Each DID requires a REGISTER. If a customer order 100 DIDs then the Ingate will
have to register 100 different accounts. The Ingate has the ability to route based on
the combination of the one REGISTER and the DID in the To Header. The To
Header of the incoming INVITE should be the AOR, or Called Number.
The Ingate dynamically Encrypts the Contact URI of the REGISTER. As per RFC,
the AOR is the To and From header and of course the Binding Address is the Contact
Header. For mostly security purposes, the Ingate creates an Encrypted URI in the
Contact Header of the Register, to ensure that no one can call the URI and prevents
Toll Fraud, and SPIT attacks and such. It also hides internal network topology, it
does provides SIP routing according to the RFC without compromising security, and
provides globally routable unique URI's.
From what was observed, the INVITE Request URI Header is perfect, calling the
Contact Header of the REGISTER. The issue is in the TO Header, currently it is the
same as the Request URI Header but this should be the AOR, or Called Number. Not
that a RFC or specification issue with this, but the ShoreTel expects to see a 10 or 11
Digit DID in this header. Here the ShoreTel can route calls to individual users or
applications. Another benefit is that you can have one REGISTER and hundreds of
DIDs in the To Header to which the IP-PBX can route the call.
This does not affect operation of the SIP Trunk, only configuration.
pg. 10
4 ShoreTel
The configuration information below shows examples for configuring the ShoreTel,
Ingate and ITSP. Even though configuration requirements can vary from setup to
setup, the information provided in these steps, along with the Planning and
Installation Guide and documentation provided by Ingate and the ITSP should prove
to be sufficient. However every design can vary and some may require more planning
then others.
4.1 ShoreTel Unsupported Features
Please consult the ShoreTel Administration Guide, Section 18 – Session Initiation
Protocol, and more specifically Section 18.1.2.5 and 18.1.2.6 for specific details
regarding unsupported features. Here is an excerpt of the Section 18.1.2.6:
General Feature Limitations
ShoreWare supports Music On Hold (MOH) over SIP trunks. The capacity
limits of MOH switches is the same as other trunks; a switch can provide
up to 15 streams. However, these streams can be to other switches or to
SIP devices.
If the ShoreTel server has a conference bridge 4.2 installed, you should not
enable SIP. The conference bridge is not compatible with a ShoreTel
system that has SIP enabled due to the dynamic RTP port required for SIP.
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.
A SIP trunk can be a member of a 3-party conference but cannot initiate a
3-way conference (unless the SIP device merges the media streams itself).
ShoreTel SIP supports basic transfers (i.e. blind transfers) and attended
transfers (i.e. consultative transfers).
Silent Monitoring is not supported on a SIP trunk call.
Barge-In is not supported on a SIP trunk call.
Call recording is not supported on a SIP trunk call. Call recording requires
presence of a physical trunk in the call.
Call redirection by SIP devices is not supported.
Park/Unpark is not supported on a SIP trunk call. This is planned for a
future release.
Silence detection on trunk-to-trunk transfers is not supported since it
requires a physical trunk.
Fax (and modem) redirection is not supported with SIP trunks as only
physical trunks can detect fax tones.
pg. 11
4.2 ShoreTel Configuration
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 Ingate SIParator.
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.
4.2.1 ShoreTel System Settings – General:
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.
4.2.1.1
Call Control Settings:
The first settings to configure within ShoreWare® Director are the Call Control
Options. To configure these settings for the ShoreTel system, log into ShoreWare
Director and select “Administration” then “Call Control” followed by “Options”
(Figure 4).
Figure 4 –Administration Call Control Options
pg. 12
The “Call Control Options” screen will then appear (Figure 5).
Figure 5 – Call Control Options
Within the “Call Control Options” SIP parameters; confirm that the appropriate
settings are made for the “Realm” “Enable SIP Session Timer” and “Always Use
Port 5004 for RTP” parameters.
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 AT&T IP
Flexible Reach Service. 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 “Callee
(UAS)”, the device of the person called will control the session timer refresh.
The next settings to verify are the “Voice Encoding and Quality of Service”,
specifically the “Media Encryption” parameter, make sure this parameter is set to
“None”, otherwise you may experience one-way audio issues. Please refer to
ShoreTel’s Administration Guide for additional details on media encryption and the
other parameters in the “Voice Encoding and Quality of Service” area.
Disabling 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 parameter is disabled,
pg. 13
Media Gateway Control Protocol (MGCP) will no longer use UDP port 5004; MGCP
and SIP traffic will use dynamic UDP ports. Once this parameter is disabled
(unchecked), make sure that “everything” (IP Phones, ShoreGear® Switches,
ShoreWare Server, Distributed Voice Mail Servers / Remote Servers, Conference
Bridges and Contact Centers) is “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.
4.2.1.2
Sites Settings:
The next settings to address are the administration of sites. These settings are
modified under the ShoreWare Director by selecting “Administration” then “Sites”
(Figure 6).
Figure 6 – Site Administration
This selection brings up the “Sites” screen. Within the “Sites” screen select the name
of the site to configure. The “Edit Site” screen will then appear. The only changes
required to the “Edit Site” screen are to the “Admission Control Bandwidth” and
“Intra-Site / Inter-Site Calls” parameters (Figure 7).
Figure 7 – Site Bandwidth settings
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.
pg. 14
4.2.1.3
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
AT&T’s IP Flexible Reach Service. Please refer to the ShoreTel Planning and
Installation Guide for additional information.
4.2.1.4
Sites Edit screen – Intra / Inter-Site Calls
By default ShoreTel 9.x has 10 built-in codecs, these codecs can be grouped as
“Codec Lists” and defined in the sites page for “Inter-site” and “Intra-site” calls.
Configure the "Inter-Site Calls" option for "Very Low Bandwidth Codecs" and save
the change. By default "Very Low Bandwidth Codecs" contains two codecs, G.729
and G.711u, with G.729 being the primary codec of choice. The “Inter-Site Calls”
parameter defines which codecs will be used when establishing a call with AT&T IP
Flexible Reach Service the preferred codec choice is G.729.
Note: Please do not modify the "Very Low Bandwidth Codecs" "Codec List".
Switch Settings - Allocating Ports for SIP Trunks
The final general settings to configure are the ShoreGear switch settings. These
changes are modified by selecting “Administration” then “Switches” followed by
“Primary” in ShoreWare Director (Figure 8).
Figure 8 – Administration Switches
pg. 15
This action brings up the “Switches” screen. From the “Switches” screen simply
select the name of the switch to configure. The “Edit ShoreGear Switch” screen will
be displayed. Within the “Edit ShoreGear Switch” screen, select the desired number
of SIP Trunks from the ports available (Figure 9).
Figure 9 – ShoreGear Switch Settings
Each port designated as a SIP Trunk Port Type enables the support for 5 individual
SIP trunks.
4.2.1.5
ShoreTel System Settings – Trunk Groups
ShoreTel Trunk Groups only support Static IP Addresses for Individual Trunks.
In trunk planning, the following needs to be considered.
-
Ingate SIParator LAN and WAN interfaces should always be configured to
use a “Static”IP Address.
The settings for Trunk Groups are changed by selecting “Administration”, then
“Trunks” followed by “Trunk Groups” within ShoreWare Director (Figure 10).
Figure 10 – Administration Trunk Groups
pg. 16
This selection brings up the “Trunk Groups” screen (Figure 11).
Figure 11 – Trunk Groups Settings
From the pull down menus on the “Trunk Groups” screen, select the site desired and
select the “SIP” trunk type to configure. Then click on the “Go” link from “Add new
trunk group at site”. The “Edit SIP Trunk Group” screen will appear (Figure 12).
Figure 12 – SIP Trunk Group Settings
The next step within the “Edit SIP Trunks Group” screen is to input the name for the
trunk group. In the example in Figure 12, the name “AT&T IP Flex Reach” has been
created. The next step is to enable (check) the setting of the “Teleworker” parameter.
Enabling the “Teleworker” parameter will configure the trunk group for inter-site
calls, which was configured earlier (see Figure 7) for “Very Low Bandwidth
Codecs” which has G.729 as the primary codec choice. Once this parameter is
enabled, it will count against the site bandwidth.
The “Enable SIP Info for G.711 DTMF Signaling” parameter should not be enabled
(checked). Enabling SIP info is currently only used with SIP tie trunks between
ShoreTel systems.
The “Profile:” parameter is new to ShoreTel 9.x and should be left at a default
setting of “_SystemTrunk”, it is not necessary to modify this parameter when
connecting to AT&T IP Flexible Reach Service via an Ingate SIParator.
pg. 17
The “Enable Digest Authentication” parameter defaults to “<None>” and
modification is not required when connecting to AT&T IP Flexible Reach Service.
The next item to change in the “Edit SIP Trunks Group” screen is to make the
appropriate settings for the “Inbound:” parameters. (Figure 13).
Figure 13 – Inbound:
Within the “Inbound:” settings, ensure the “Number of Digits from CO” is set to
match what the ShoreGear SIP trunk switch will be receiving from AT&T IP Flexible
Reach Service and ensure the “DNIS” or “DID” box is enabled (checked), along with
the Extension parameter. We recommend that the Tandem Trunking parameter be
enabled (checked) otherwise transfers to external telephone numbers will fail via SIP
trunks. For additional information on this parameter please refer to ShoreTel’s
Planning and Installation Guide.
Note: Geos Communications can be provisioned to deliver 10, 7 or 4 digits for
incoming calls, the ShoreTel Trunk Group will only allow one setting. If your service
has been provisioned for 10, 7 or 4 digits configure the trunk group for 10 digits, you
will then need to configure Ingate to add digits to the 7 or 4 digit calls, so that they
will be 10 digits in length. See section 5 Configuring Ingate to convert 7 or 4 digit
called number to 10 digits.
The following section is configured in the same way as any normal Trunk Group.
pg. 18
Figure 14 – Outbound and Trunk Services:
Enable (check) the “Outbound” parameter and define a Trunk “Access Code”
and “Local Area Code” as appropriate.
In the “Trunk Services:” area, make sure the appropriate services are enabled or
disabled based on what AT&T IP Flexible Reach Service supports and what features
are needed from this Trunk Group.
The last parameter determines if the call is sent out as <unknown> or with caller
information (Caller ID). User DID will impact how information is passed out to the
SIP Trunk group.
After these settings are made to the “Edit SIP Trunk Group” screen, select the “Save”
button to input the changes.
The last configuration parameters for configuration in the Trunk Group is “Trunk
Digit Manipulation” (figure 15):
pg. 19
Figure 15 – Trunk Digit Manipulation:
The only parameter that needs adjustment (from default) to interface with AT&T IP
Flexible Reach Service is “Dial 7 digits for Local Area Code”, disable (uncheck)
this parameter. Save the changes.
This completes the settings needed to set up the trunk groups on the ShoreTel system.
ShoreTel 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
(Figure 16).
Figure 16 – Individual Trunks
The “Trunks by Group” screen is used to change the individual trunks settings that
appear (Figure 17).
Figure 17 – Trunks by Group
pg. 20
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 “AT&T IP Flex Reach”, as
created above, see Figure 12. Click on the ”Go” button to bring up the “Edit Trunk”
screen (Figure 18).
Figure 18 - Edit Trunks Screen for Individual Trunks
From the individual trunks “Edit Trunk” screen, input a name for the individual trunks,
select the appropriate switch, select the SIP Trunk type and input the number of
trunks. When selecting a name, the recommendation 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. For the “Sip Trunk
Type”, the only option available is “Use IP Address”, define the IP address of the
Ingate SIParator product. The last step is to select the number of individual trunks
desired (each one supports “one” audio path – example if 10 is configured, then 10
audio paths can be up at one time). Once these changes are complete, select the “Save”
button to commit changes.
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 Ingate SIParator – just the
same as if PRI were being used.
After setting up the trunk groups and individual trunks, refer to the ShoreTel Planning
and Installation Guide to make the appropriate changes for the User Group settings.
This completes the settings for the ShoreTel system side.
pg. 21
5 Ingate
Ingate Systems AB is a Stockholm, Sweden based high-tech company that designs,
develops, manufactures and markets leading data communications products for trusted
Unified Communications. Ingate designed the world’s first Session Initiation Protocol
(SIP)-capable firewalls and SIParators, products that enable Unified Communications
over the Internet.
Unified Communications, with applications such as Internet telephony, presence
indication, instant messaging, and audio/video conferencing, are modern and powerful
business tools that enable enterprises to maintain reliable IP-communications
internally and externally. As more businesses utilize these applications, service
providers are offering SIP trunks to connect Local Area Networks to the outer world
via Internet and/or dedicated, managed IP-lines.
The enterprise Session Border Controller (Firewall) needs to manage all incoming and
outgoing traffic securely. Authorized traffic based on SIP needs to pass through the
Session Border Controller in a controlled manner reaching SIP units inside and
outside the LAN. Ingate's Session Border Controllers are compatible with existing
networks, and allow businesses to utilize the cost and time saving benefits of IP-based
real-time communications with minimum investment.
Ingate’s leading products are marketed through world leading distributors, Value
Added resellers and OEM’s on all continents.
5.1 Ingate Product Information
Ingate SIParator and Firewall products are compatible with communications
equipment from other vendors and service providers who support the SIP Protocol.
The Ingate products are a security device designed to sit on the Enterprise network
edge, an ICSA Labs Certified security product, focused on SIP communications
security and network security for the Enterprise.
Ingate products are designed to solve the issues related to SIP traversing the NAT
(Network Address Translation) which is a part of all enterprise class firewalls. The
NAT translates between the public IP address(es) of the enterprise, and the private IP
addresses which are only known inside the LAN. These private IP addresses are
created to enable all devices to have an IP address, and also provide one of the
security layers of the enterprise network.
In addition, the Ingate products provide routing rules to flexibility in SIP traffic flows
and ensure only allowed SIP traffic will pass. This provides an ability to route any
call to any destination in a secure manner.
The Ingate products also contain SIP Protocol normalization tools to assist in the
interoperability of all of the different SIP vendors and service providers. Features
such as a B2BUA and other advanced customization tools allow integration with any
other vendor.
pg. 22
5.2 Ingate Product Configuration
The following section will briefly describe the configuration of the Ingate products.
Further configuration of the Ingate products can be found under the Account Login
page. http://www.ingate.com/Login.php Including a Configuration Guide for the
Ingate with a ShoreTel when using SIP Trunking.
5.2.1 Startup Tool
The Ingate Startup Tool is an installation tool for Ingate Firewall® and Ingate
SIParator® products, facilitates the “out of the box” set up of SIP Trunking solutions
with ShoreTel and various Internet Telephony Service Providers. Designed to
simplify SIP trunk deployments, the tool will automatically configure a user’s Ingate
Firewall or SIParator® to work with ShoreTel and the SIP Trunking service provider
of your choice. With the push of a button, the configuration tool will automatically
create a SIP trunk deployment designed to the user’s individual setup.
Users can select ShoreTel from a drop-down menu and the Internet Telephony
Service Provider (ITSP) they use; the configuration tool will automatically apply the
correct settings to the Ingate Firewall or SIParator to work seamlessly with that
vendor or service provider. A list of SIP Trunking service providers that have
demonstrated interoperability with the Ingate products is incorporated into the
interface. Please note that not all SIP Trunking service providers listed in this
interface have been certified by ShoreTel. Consult the ShoreTel Certified Technology
Partner list of vendors for a current list.
(http://www.shoretel.com/partners/technology/certified_partners.html)
The configuration tool is available now as a free download for all Ingate Firewalls and
SIParators. It can be found at http://www.ingate.com/Startup_Tool.php. Also
available here is a Startup Tool Getting Started Guide to assist in using the Startup
Tool.
pg. 23
5.2.1.1
Contacting the Ingate Unit
There are three main options to keep in mind. 1) Is this an “Out of the Box”
installation, if so select “Configure the unit for the first time.” 2) If the Ingate has a
configuration already, then select “Change or update configuration of the unit”. And
3) Select “Configure SIP Trunking” to have the available options for SIP Trunking.
Select one or the
other
Select Configure SIP
Trunking
Assign IP Addresses
and MAC the tool
will use to config the
Ingate
Assign Password the
tool will use to config
the Ingate
Status Information, helpful
for troubleshooting
pg. 24
5.2.1.2
Startup Tool Configuration – Network Topology
The Network Topology tab is about defining how the Ingate product will be deployed
and the Network Topology required around it. The Product Type defines the
deployment and the rest define the IP Addresses and Masks and DNS Servers.
Select the deployment
according to the picture
Assign IP Addresses,
the tool will config
the Ingate
for troubleshooting
pg. 25
5.2.1.3
Startup Tool Configuration – IP-PBX
Selecting ShoreTel as the IP-PBX Type will ensure the Ingate Startup Tool
propagates the necessary configuration into the Ingate SIParator or Firewall to ensure
correct operation. This configuration is based off of extensive TTP Interop Testing as
well as experience in various deployments. Simply assign the IP Address of the
ShoreTel ShoreGear Switch.
Assign the ShoreTel
as the IP-PBX Type
Assign the IP
Address of the
ShoreTel ShoreGear
for troubleshooting
pg. 26
5.2.1.4
Startup Tool Configuration – ITSP
Ingate has gone out to verify and test with a large number of Carriers and Service
Providers. With every ITSP verification, Ingate records the individual setup and
deployment characteristics of each Service Provider. With a simple selection of the
Service Provider the knowledge and configuration of each deployment is propagated
to the Ingate SIParator or Firewall.
Select ITSP Vendor
Assign the ITSP IP
Address
User Account Information,
DID Assignment and
Registration Authentication
for troubleshooting
pg. 27
5.2.1.5
Upload Configuration
At this point the Startup Tool has all the information required to push a database into
the Ingate unit. The Startup Tool can also create a backup file for later use.
1. Press the “Upload” button. If you would like the Startup Tool to create a
Backup file also select “Backup the configuration”. Upon pressing the
“Upload” button the Startup Tool will push a database into the Ingate unit.
2. When the Startup has finished uploading the database a window will appear
and once pressing OK the Startup Tool will launch a default browser and
direct you to the Ingate Web GUI.
pg. 28
3. Although the Startup Tool has pushed a database into the Ingate unit, the
changes have not been applied to the unit. Press “Apply Configuration” to
apply the changes to the Ingate unit.
4. A new page will appear after the previous step requesting to save the
configuration. Press “Save Configuration” to complete the saving process.
pg. 29
5.3 Ingate Troubleshooting
5.3.1 Call Flow Examples
5.3.1.1
Incoming Call
Incoming calls will always originate from the Service Provider and be addressed
directly to the Ingate SIParator IP Address. The Ingate in turn will route the call to
the ShoreGear switch.
5.3.1.2
Outgoing Call
Outgoing calls will always originate from the ShoreTel Phones, then the ShoreGear
switch makes a call directly to the Ingate IP address. The Ingate in turn will route the
call to the ITSP.
pg. 30
5.3.2 Startup Tool
5.3.2.1
Status Bar
Located on every page of the Startup Tool is the Status Bar. This is a display and
recording of all of the activity of the Startup Tool, displaying Ingate unit information,
software versions, Startup Tool events, errors and connection information. Please
refer to the Status Bar to acquire the current status and activity of the Startup Tool.
5.3.2.2
Configure Unit for the First Time
Right “Out of the Box”, sometimes connecting and assigning an IP Address and
Password to the Ingate Unit can be a challenge. Typically, the Startup Tool cannot
program the Ingate Unit. The Status Bar will display “The program failed to assign
an IP address to eth0”.
Possible Problems
Ingate Unit is not Turned On.
Ethernet cable is not connected to Eth0.
Incorrect MAC Address
An IP Address and/or Password have already
been assigned to the Ingate Unit
Ingate Unit on a different Subnet or Network
Possible Resolution
Turn On or Connect Power
(Trust me, I’ve been there)
Eth0 must always be used with the Startup Tool.
Check the MAC address on the Unit itself. MAC Address
of Eth0.
It is possible that an IP Address or Password have been
already been assigned to the unit via the Startup Tool or
Console
The Startup Tool uses an application called “Magic PING”
to assign the IP Address to the Unit. It is heavily reliant on
ARP, if the PC with the Startup Tool is located across
Routers, Gateways and VPN Tunnels, it is possible that
MAC addresses cannot be found. It is the intension of the
Startup Tool when configuring the unit for the first time to
pg. 31
Possible Problems
Despite your best efforts…
Possible Resolution
keep the network simple. See Section 3.
1. Use the Console Port, please refer to the Reference
Guide, section “Installation with a serial cable”, and
step through the “Basic Configuration”. Then you
can use the Startup Tool, this time select “Change or
Update the Configuration”
2. Factory Default the Database, then try again.
5.3.2.3
Change or Update Configuration
If the Ingate already has an IP Address and Password assigned to it, then you should
be able use a Web Browser to reach the Ingate Web GUI. If you are able to use your
Web Browser to access the Ingate Unit, then the Startup should be able to contact the
Ingate unit as well. The Startup Tool will respond with “Failed to contact the unit,
check settings and cabling” when it is unable to access the Ingate unit.
Possible Problems
Ingate Unit is not Turned On.
Incorrect IP Address
Incorrect Password
Despite your best efforts…
Possible Resolution
Turn On or Connect Power
Check the IP Address using a Web Browser.
Check the Password.
1. Since this process uses the Web (http) to access
the Ingate Unit, it should seem that any web
browser should also have access to the Ingate
Unit. If the Web Browser works, then the
Startup Tool should work.
2. If the Browser also does not have access, it
might be possible the PC’s IP Address does not
have connection privileges in “Access Control”
within the Ingate. Try from a PC that have
access to the Ingate Unit, or add the PC’s IP
Address into “Access Control”.
pg. 32
5.3.2.4
Network Topology
There are several possible error possibilities here, mainly with the definition of the
network. Things like IP Addresses, Gateways, NetMasks and so on.
Possible Problems
Error: Default gateway is not reachable.
Error: Settings for eth0/1 is not correct.
Error: Please provide a correct netmask for
eth0/1
Error: Primary DNS not setup.
Possible Resolution
The Default Gateway is always the way to the
Internet, in the Standalone or Firewall it will be the
Public Default Gateway, on the others it will be a
Gateway address on the local network.
IP Address of Netmask is in an Invalid format.
Netmask is in an Invalid format.
Enter a DNS Server IP address
5.3.2.5
IP-PBX
The errors here are fairly simple to resolve. The IP address of the IP-PBX must be on
the same LAN segment/subnet as the Eth0 IP Address/Mask.
Possible Problems
Error: The IP PBX IP does not seem to be on
the LAN.
Error: You must enter a SIP domain.
Error: As you intend to use RSC you must
enter a SIP domain. Alternatively you may
configure a static IP address on eth1 under
Network Topology
Possible Resolution
The IP Address of the IP-PBX must be on the same
subnet as the inside interface of the Ingate Eth0.
Enter a Domain, or de-select “Use Domain”
Enter a Domain or IP Address used for Remote SIP
Connectivity. Note: must be a Domain when used
with SIP Trunking module.
pg. 33
5.3.2.6
ITSP
The errors here are fairly simple to resolve. The IP address, Domain, and DID of the
ITSP must be entered.
Possible Problems
Error: Please enter a domain name for your
provider
Error: Please enter number, name and domain.
5.3.2.7
Possible Resolution
Enter a Domain, or de-select “Use Domain”
Enter a DID and Domain, or de-select “Use
Account”
Apply Configuration
At this point the Startup Tool has pushed a database to the Ingate Unit, you have
Pressed “Apply Configuration” in Step 3) of Section 4.7 Upload Configuration, but
the “Save Configuration” is never presented. Instead after a period of time the
following webpage is presented. This page is an indication that there was a change in
the database significant enough that the PC could no longer web to the Ingate unit.
Possible Problems
Eth0 Interface IP Address has changed
Access Control does not allow administration
from the IP address of the PC.
Possible Resolution
Increase the duration of the test mode, press “Apply
Configuration” and start a new browser to the new
IP address, then press “Save Configuration”
Verify the IP address of the PC with the Startup
Tool. Go to “Basic Configuration”, then “Access
Control”. Under “Configuration Computers”,
ensure the IP Address or Network address of the PC
is allowed to HTTP to the Ingate unit.
pg. 34
5.3.3 Ingate Example Configuration
Configure your Ingate Firewall or Ingate SIParator to get basic network connectivity
on all applicable interfaces. Please refer to the Reference Guide and other
documentation as needed.
Remember to configure the following:
Assign IP addresses on the inside and outside interface. For DMZ SIParators,
use one interface only. (Network -> All Interfaces)
Assign a default gateway. (Network -> Default Gateway)
Assign a DNS server address. (Basic Configuration -> Basic Configuration)
Define the IP subnet allowed to configure the Ingate and the interfaces to use
for configuration. (Basic Configuration -> Access Control)
First make these basic settings and apply the configuration to have the unit working in
your network environment. Then proceed with the following settings to get SIP
Trunking to work with your service provider.
5.3.3.1
Network and Computers
This is an example of the Network – Networks and Computers page with an Ingate
SIParator in a Stand-alone configuration. The Networks and Computer page is a IP
Table List or Route List, providing the Ingate knowledge of its surrounding networks
and what interface they are connected too. Also, the table provides identification of
specific IP Addresses for later use in providing filter and identification of source IP
addresses in the Dial Plan and other locations.
Add a network for the Service Provider (ITSP IP). If you don’t know the IP
addresses used for the ITSP, you can put in 0.0.0.0 as lower limit and
255.255.255.255 as upper limit. In this way, requests from any IP address will
be accepted.
Add a network for the LAN (inside IP range).
Add an IP Address of the ShoreTel ShoreGear
pg. 35
5.3.3.2
Basic Configuration – SIParator Type (SIParator
Only)
Use the appropriate SIParator configuration for your deployment.
5.3.3.3
SIP Traffic – Filtering
Under Proxy Rules, change the Default Policy for SIP Requests to “Process
All”.
Content Type: Add */* and Allow - ON
pg. 36
5.3.3.4
Interoperability
There are some general Interop settings required for use with ShoreTel.
Configuration Steps:
URI Encoding – Use shorter, encrypted URI
Signaling Order of Re-INVITEs – Send response before re-INVITE are
forwarded
Allow Large UDP Packets - Allow Large UDP Packets
pg. 37
5.3.3.5
Dial Plan
This is an example of the SIP Traffic - Dial Plan on the Ingate SIParator. There are
three main parameters that need to be defined to create the Dial Plan. Matching From
Header, Matching Request URI and Forward To are parameters that when combined
together form the Dial Plan.
The key difference in the ShoreTel integration is the use of the B2BUA as the SIP
Normalization Tool, thus the Forward To section Regular Expression has
sip:[email protected];b2bua for the ShoreTel ShoreGear destination as well as
sip:[email protected];b2bua for the Service Provider destination.
pg. 38
5.3.4 Ingate Troubleshooting Tools
5.3.4.1
Display Logs
Here is the internal logging of the Ingate. The Display Logs show all SIP Signaling
and also TLS (SSH) certificate exchange and setup.
Press “Display Log”
to see internal logs
Always create a
“Support Report”
for Ingate Support
Show newest
log on top
Filter on SIP
specific fields
Filter on SIP
traffic only
pg. 39
5.3.4.2
Packet Capture
The Packet Capture capability of the Ingate allows for the capture and export of all
traffic on any one or ALL interfaces simultaneously. Then export to your PC where it
can be viewed in Wireshark or Ethereal.
Select “All Interfaces” to
cook multiple captures
from multiple interfaces
into one PCAP
Filter on Port,
Transport and
other criteria
Download PCAP File
Start Capture,
reproduce the
problem, then
Stop Capture
5.3.4.3
Check Network
Standard PING and Trace Route feature for simple network checks.
PING and Trace Route
pg. 40
pg. 41
5.4 Ingate Sales & Technical Support
5.4.1 Sales
North America
For general sales questions or resellers who want to start selling this solution, please
contact; Steven Johnson 603-883-6569 or [email protected]
www.ingate.com
EMEA
For general sales questions or resellers who want to start selling this solution please
contact; Ingate Systems HQ +46-8-600-7750 or [email protected]
www.ingate.com
5.4.2 Technical Support
North America Customers:
The Ingate Authorized Reseller should always be your first contact for support. The
ShoreTel TAC is a part of the Ingate Authorized Resellers. If you don't work with an
Ingate Authorized Reseller, you may purchase an Annual Support Agreement from
Ingate Systems.
All support questions and issues should be directed to:
Email: [email protected]
Phone: 1-866-809-0002
Operational Hours: 8:00am to 6:00pm EST
EMEA Customers:
The Ingate Authorized Reseller should always be your first contact for support. The
ShoreTel TAC is a part of the Ingate Authorized Resellers. If you don't work with an
Ingate Authorized Reseller, you may purchase an Annual Support Agreement from
Ingate Systems.
All support questions and issues should be directed to:
Email: [email protected]
Phone old: +46-13-21 08 52
Phone new: +46-8 600 7766
Operational Hours: 8:00am - 5:00pm CET (UTC+1)
pg. 42
6 Document and Software Copyrights
Copyright © 2009 by ShoreTel, Inc., Sunnyvale, California, U.S.A. All rights
reserved. Printed in the United States of America. Contents of this publication may
not be reproduced or transmitted in any form or by any means, electronic or
mechanical, for any purpose, without prior written authorization of ShoreTel
Communications, Inc.
ShoreTel, Inc. reserves the right to make changes without notice to the specifications
and materials contained herein and shall not be responsible for any damage (including
consequential) caused by reliance on the materials presented, including, but not
limited to typographical, arithmetic or listing errors.
7 Trademarks
The ShoreTel logo, ShoreTel, ShoreCare, ShoreGear, ShoreWare and ControlPoint
are registered trademarks of ShoreTel, Inc. in the United States and/or other countries.
ShorePhone is a trademark of ShoreTel, Inc. in the United States and/or other
countries. All other copyrights and trademarks herein are the property of their
respective owners.
8 Disclaimer
To be “ShoreTel Certified” means that Technology Partner's product will interoperate
with the ShoreTel system, but ShoreTel does not certify that the features or
functionality of Technology Partner's product will perform as specified by
Technology Partner nor that Technology Partner's product will meet your specific
application needs or requirements. To inter-operate means that Technology Partner's
product is able to exchange, use and share information with the ShoreTel system.
9 Company Information
ShoreTel, Inc.
960 Stewart Drive
Sunnyvale, California 94085 USA
+1.408.331.3300
+1.408.331.3333 fax
pg. 43

Geos Communications SIP Trunking

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