ShoreTel Tele2 SIP Trunking (TPP 13001)

I n n o v a t i o n
A p p N o t e
N e t w o r k
TPP- 13001
Date : June, 2013
Product: ShoreTel | Tele2
System version: ShoreTel 13.x
ShoreTel & Tele2 (Netherlands) for SIP
Trunking (Native)
SIP trunking allows the use of Session Initiation Protocol (SIP) communications from Tele2 instead of
traditional analog, Basic Rate Interface (BRI), T-1 or E-1 trunk connections.
This application note provides details on connecting the ShoreTel Unified Communications IP phone
system to Tele2 for SIP trunking.
Table of Contents
Table of Contents .................................................. 1
Overview............................................................... 2
Tele2 Overview..................................................... 2
Tele2’s Native SIP Trunk Deployment Topology . 2
Version Support .................................................... 3
SIP Certification Testing Results – Tele2 .............. 3
SIP Certification Testing Results – ShoreTel......... 8
ShoreTel Unsupported Features .......................... 9
General Feature Limitations ....................................9
ShoreTel Configuration ...................................... 10
Configuring Call Control Settings .........................10
Additional Details...................................................12
ShoreTel Site Settings ............................................14
Configuring Switch Settings for SIP Trunks ...........15
Creating a SIP Profile .............................................17
Creating the SIP Trunk Group ...............................18
DDI/DID Assignment .............................................20
Creating Individual Trunks ....................................21
Custom Dial Strings for Tele2 ................................23
Re-routing Emergency Numbers ..........................24
Customizing CallerID on Forwarded Calls ...........27
Conclusion .......................................................... 29
Document & Software Copyrights ..................... 29
Trademarks ........................................................ 29
Disclaimer........................................................... 29
Company Information ........................................ 29
ShoreTel tests and validates the interoperability of the Member's solution with ShoreTel's published software
interfaces. ShoreTel does not test, nor vouch for the Member's development and/or quality assurance
process, nor the overall feature functionality of the Member's solution(s). ShoreTel does not test the
Member's solution under load or assess the scalability of the Member's solution. It is the responsibility of the
Member to ensure their solution is current with ShoreTel's published interfaces.
The ShoreTel Technical Support organization will provide Customers with support of ShoreTel's published
software interfaces. This does not imply any support for the Member's solution directly. Customers or reseller
partners will need to work directly with the Member to obtain support for their solution.
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 1
Overview
This document provides details for connecting the ShoreTel® system to the Tele2-based network for
SIP trunking enabling audio communications between the ShoreTel UC system and the Public
Switched Telephone Network (PSTN). This document also discusses the network architecture
needed to set up these systems to interoperate.
Tele2 Overview
Tele2 is one of Europe’s leading telecom operators, offering mobile services, fixed broadband and
telephony, data network services, cable TV and content services. Tele2 has 38 million customers in
11 countries.
Tele2 offers fixed telephony services for businesses in seven countries offering both IP telephony
solutions and traditional fixed telephony.
Web sites: www.tele2.com / www.tele2.nl
Tele2’s Native SIP Trunk Deployment Topology
The Tele2 deployment of native SIP trunks does not require a Session Border Controller to be
located at the customer’s premise. This implies that the customer has agreed to allow unfiltered and
non-firewalled connectivity between the customers LAN and the Tele2 network.
Security in a SIP environment is very important. ShoreTel always recommends that the customer
protect their networks from SIP-related threats. An effective security strategy is the deployment of a
SIP-specific Session Border Controller (SBC) that is installed on-premise at the customer’s site and
provides SIP protocol filtering and SIP packet inspection as a Back-to-Back User Agent (B2BUA).
Other security alternatives can include SIP Application Layer Gateways and/or SIP-based Firewall
device.
By choosing to deploy “raw”, or native, SIP trunking with Tele2 (with no on-premise SIP SBC, ALG or
firewall) the customer is agreeing to “outsource” their SIP security to Tele2.
At the time of this writing the Tele2 SIP network utilizes the following components:
 Acme Packets (now owned by Oracle) Net-Net 4500 Session Border Controller
 Genband CS2000 Soft Switch
Do we have a DIAGRAM to insert to show the connected pieces
In this deployment topology, all of the ShoreTel devices need to be able to route IP packets directly to
and from the IP address of the Tele2 Acme Packets SBC. This includes all IP Phones, all
ShoreGear voice appliances, all ShoreTel servers and any computers (PCs and Macs) on the
network that will use the ShoreTel Communicator software as a SoftPhone.
NOTE:
Additional IP routes may need to be added to networking equipment at the customer’s premise
to ensure the proper routing of packets to the Tele2 SBC.
NOTE:
Testing with Tele2’s Native SIP Trunk offering was performed with no customer-side SBCs, NAT
devices, firewalls or ALGs.
In our test environment we had a Tele2 router on-site in addition to the customer’s standard Internet
gateway. All computers on the network were configured with the Internet Gateway as their default
gateway for IP routing purposes. We needed to add a static host route on the Internet Gateway
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 2
identifying the Tele2 SBC’s IP address and redirecting traffic to the Tele2 Router. This would be
used by any computer that initiates their SoftPhone on their computer. The Internet Gateway would
redirect the traffic back through the Tele2 router.
Additionally, we manually configured all ShoreTel hardware including IP Phones and ShoreGear
switches with a default gateway of the Tele2 Router, not the Internet Gateway. This prevents the IP
Phones and ShoreGear switches from being overloaded with ICMP Redirect traffic from the Internet
Gateway.
Finally, on the ShoreTel Server(s) we added a persistent host route to the Tele2 SBC’s IP address
pointed at the Tele2 router.
Can we screen shot the route add?
NOTE:
Consult with a network professional for assistance with your IP Routing configuration.
Version Support
Member products are certified via the ShoreTel Innovation Network Certification Process.
Testing for this ITSP was done using the following versions:
ShoreTel Release
v. 13.1
Tele2’s Acme Packet 4500 SBC
v. 6.3.7
SIP Certification Testing Results – Tele2
Tele2-specific SIP trunk certification tests performed:
ID
1
Name
Sip01
2
3
4
5
6
Callp11a
Callp11b
Callp12a
Callp12b
Media01a
7
Media01b
8
Callp09b
9
Callp09c
10
Callp10b
Title
SIP heartbeat
Description
30 sec interval SIP OPTIONS Ping message activates
circuit
* ShoreTel sends SIP Options Ping packets that are received and responded to by
the Tele2 SBC. This exchange of packets allows both ends to maintain “keepalive” and session status. The Tele2 SBC also sends SIP Options Ping packets
to the ShoreTel PBX. ShoreTel does not support the format of those Ping
packets at this time.
Forward Release 1
PBX phone initiates the call
Forward Release 2
PSTN subscriber initiates the call
Backward Release 1
PBX phone initiates the call
Backward Release 2
PSTN subscriber initiates the call
Early media transport before Call from PBX phone to PSTN subscriber. Ring back
connect, PBX to PSTN
tone and “183 Session Progress” or “180 Ringing”
message.
Early media transport before Call from PSTN subscriber to PBX phone. Ring back
connect, PSTN to PBX
tone and “183 Session Progress” or “180 Ringing”
message.
Busy Number, PBX to
Call from PBX phone to PSTN subscriber. “486 Busy”
PSTN subscriber
message.
Busy Number, PSTN to
Call from PSTN subscriber to PBX phone. “486 Busy”
PBX
message.
Invalid number, PBX to
PBX dials non-existing number. Network returns “404
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Result
Pass*
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Page 3
PSTN subscriber
Invalid number, PSTN to
PBX
Not Found” message.
Destination DDI/DID ‘X’ is within proper DDI/DID range
assigned to PBX, but not allocated. PBX returns “404
Not Found” message.
PBX releases the call before PSTN answers the call.
11
Callp10c
12
Callp13a
13
Callp13b
14
Callp13c
15
Callp13d
16
Callp14a
Early Call Termination, PBX
to PSTN
Early Call Termination,
PSTN to PBX
Early Call Termination on
tariff announcement
Early Call Termination, T9
timer expiry
No Reply, T9 timer expiry 1
17
Callp14b
No Reply, T9 timer expiry 2
18
Callp14c
19
Callp14d
20
Max01a
21
Max01b
22
23
24
Short01a
Long01a
Callp23/24a
25
Callp23/24b
26
Callp23/24c
27
28
29
30
31
32
33
34
35
Callp25a
Callp25b
Callp25c
Callp25d
Callp26a
Callp26b
xferA01
xferA03
xferA07
No Answer, PBX calls
PSTN
No Answer, PSTN calls
Network releases the call with CANCEL (cause reason
PBX
“Recovery of Timer Expiry”).
Exceed max. SIP trunk
Network returns “503 Service Unavailable”.
sessions, PBX to PSTN
* Tele2 expects to see a call placed when the trunk maximum is exceeded so they
can respond with a “503 Service Unavailable” message. Instead, ShoreTel never
exceeds the max trunk limit and plays fast re-order tone to an internal caller when
all trunks are in use.
Exceed max. SIP trunk
Network returns “34 No channel available”.
sessions, PSTN to PBX
PBX to PSTN, short call
Successful call with duration <1 second.
PBX to PSTN, long call
Successful call with duration > 4 hours.
Call Forward Always, PSTN Successful call with proper FROM and DIVERSION
to PBX to PSTN
headers.
* Call is placed properly and successfully. The formatting of the information in the
SIP packet is valid but not as expected by Tele2
Call Forward Busy, PSTN to Successful call with proper FROM and DIVERSION
PBX to PSTN
headers.
* Call is placed properly and successfully. The formatting of the information in the
SIP packet is valid but not as expected by Tele2
Call Forward No Answer,
Successful call with proper FROM and DIVERSION
PSTN to PBX to PSTN
headers.
* Call is placed properly and successfully. The formatting of the information in the
SIP packet is valid but not as expected by Tele2
Call Hold, No Music 1
PBX calls PSTN, PBX initiates Hold
Call Hold, No Music 2
PBX calls PSTN, PSTN initiates Hold
Call Hold, No Music 3
PSTN calls PBX, PBX initiates Hold
Call Hold, No Music 4
PSTN calls PBX, PSTN initiates Hold
Call Hold, with MoH 1
PSTN calls PBX, PBX initiates Hold
Call Hold, with MoH 2
PBX calls PSTN, PSTN initiates Hold
Consultative Transfer 1
PSTN calls PBX, Consult Xfer to PSTN
Consultative Transfer 2
PBX user 1 calls PBX user 2, Consult Xfer to PSTN
Consultative Transfer 3
PSTN calls PBX user 1, Consult Xfer to PBX user 2
Pass
Pass
PSTN releases the call before PBX answers the call.
Pass
PBX releases the call after tariff announcement. Call
duration listed as ‘0 sec.’
T9 timer expires and network releases the call.
Pass
T9 timer expires before SIP T1 and calling party
network releases the call.
T9 timer expires before SIP T1 and calling party
network releases the call.
PBX receives “480 Temporarily unavailable” message.
N/A
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Pass
N/A
Pass
Pass
Pass*
Pass
Pass
Pass
Pass*
Pass*
Pass*
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Page 4
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
xferB01
xferB03
xferB07
trans01a
Blind Transfer 1
PSTN calls PBX, Blind Xfer to PSTN
Blind Transfer 2
PBX user 1 calls PBX user 2, Blind Xfer to PSTN
Blind Transfer 3
PSTN calls PBX user 1, Blind Xfer to PBX user 2
Support for UDP based SIP PSTN calls PBX
signaling 1
trans01b
Support for UDP based SIP PBX calls PSTN
signaling 2
trans02a
Support for TCP based SIP PSTN calls PBX
signaling 1
* ShoreTel does not support SIP over TCP.
trans02b
Support for TCP based SIP PBX calls PSTN
signaling 2
* ShoreTel does not support SIP over TCP.
Reach01
National call, starting with
INVITE from PBX shows
+31[123457]
sip:+31xxxxxyyyy@<domain>
Reach02
International call, starting
INVITE from PBX shows sip:+CCxxxxyyyy@<domain>
with +CC
Reach03
Mobile call, starting with
INVITE from PBX shows
+316
sip:+316xxxxyyyy@<domain>
Reach04
Free phone call, starting
INVITE from PBX shows sip:+31800xxxx@<domain>
with +31800
Reach05a
premium rate call, starting
INVITE from PBX shows sip:+31900xxxx@<domain>
with +31900
Reach06b
premium rate call, starting
INVITE from PBX shows sip:+31906xxxx@<domain>
with +31906
Reach06c
premium rate call, starting
INVITE from PBX shows sip:+31909xxxx@<domain>
with +31909
Reach07
directory inquiry call,
INVITE from PBX shows
+3114001802
sip:+3114001802@<domain>
Reach08
directory inquiry call,
INVITE from PBX shows
+311400116000
sip:+311400116000@<domain>
Reach09
directory inquiry call,
INVITE from PBX shows
+31140014020
sip:+31140014020@<domain>
Reach10a/b Emergency call to
INVITE from PBX shows
+311412PE112
sip:+31141210112@<domain>
g3fax01a
Group 3 Fax, pass-through
10 page fax test. Verify that fax detection is performed
mode, PBX fax to PBX fax
on both sides and completed within 15 minutes.
g3fax01b
Group 3 Fax, pass-through
10 page fax test. Verify that fax detection is performed
mode, PBX fax to PSTN fax on both sides and completed within 15 minutes.
g3fax01c
Group 3 Fax, pass-through
10 page fax test. Verify that fax detection is performed
mode, PSTN fax to PBX fax on both sides and completed within 15 minutes.
t38fax01a
Group 3 Fax, T.38 relay
10 page fax test. Verify that fax detection is performed
mode, PBX fax to PSTN fax on both sides that T.38 is used and fax transmission
completed within 15 minutes.
t38fax01b
Group 3 Fax, T.38 relay
10 page fax test. Verify that fax detection is performed
mode, PSTN fax to PBX fax on both sides that T.38 is used and fax transmission
completed within 15 minutes.
t38fax01c
Group 3 Fax, T.38 relay
10 page fax test. Verify that fax detection is performed
mode, PBX fax to PBX fax
on both sides that T.38 is used and fax transmission
completed within 15 minutes.
t38fax01d
Group 3 Fax, T.38 relay
PBX rejects T.38 request. Faxing proceeds using
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Pass
Pass
Pass
Pass
Pass
Fail*
Fail*
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Pass
N/A
Pass
Pass
Pass
Pass
N/A
N/A
Page 5
61
sg3fax01a
62
sg3fax01b
63
sg3fax01c
64
Codec01a
65
Codec01b
66
Dtmf01a
67
Dtmf01b
68
Scr01a
69
Scr01b
70
Scr01c
71
Clip1a
72
Clir01a
73
Clir01c
74
Robust01a
75
Robust01b
76
Robust01c
77
Robust02a
78
Robust02b
79
Auth03a
mode fallback to G.711,
PBX fax to PBX fax
Super G3 fax/modem, passthrough mode, PBX fax to
PBX fax
Super G3 fax/modem, passthrough mode, PBX fax to
PSTN fax
Super G3 fax/modem, passthrough mode, PSTN fax to
PBX fax
PBX to PSTN subscriber,
G.711Alaw and RFC2833
PSTN subscriber to PBX,
G.711Alaw and RFC2833
PBX to PSTN DTMF
transparency using RFC
2833
PSTN to PBX DTMF
transparency using RFC
2833
CLI number on outbound
calls, valid DDI/DID
CLI number on outbound
calls, invalid DDI/DID
CLI number on outbound
calls, no number sent
CLI number on inbound
calls, CLI permitted
CLI number on outbound
calls, CLI restricted
CLI number on inbound
calls, CLI restricted
Service recovery after long
loss of connectivity, no
active call
Service recovery after short
loss of connectivity, active
call
Service recovery after long
loss of connectivity, active
call
Service recovery after loss
of connectivity, no active
calls
Service recovery after loss
of connectivity, active calls
PBX to PSTN, source IP
G.711.
10 page fax test. Verify that fax detection is performed
on both sides that T.38 is used and fax transmission
completed within 15 minutes.
10 page fax test. Verify that fax detection is performed
on both sides that T.38 is used and fax transmission
completed within 15 minutes.
10 page fax test. Verify that fax detection is performed
on both sides that T.38 is used and fax transmission
completed within 15 minutes.
Verify that RTP has G.711Alaw payload.
Verify bidirectional transport of DTMF.
Verify that RTP has G.711Alaw payload.
Verify bidirectional transport of DTMF.
Verify that DTMF events are sent and received using
RTP DTMF event packets
N/A
Verify that DTMF events are sent and received using
RTP DTMF event packets
Pass
Verify proper CallerID number information in FROM or
P-Asserted-Identity fields.
Verify that CLI screening announcement is offered to
PBX
Verify that CLI screening announcement is offered to
PBX
Verify CLI number is seen in INVITE to PBX
Pass
Verify CLI number sent by PBX is reset to
“Anonymous” by network when calling PSTN
Verify that the INVITE request to PBX contains
anonymous ‘from’ header and ‘privacy=id’ header
Loss of circuit drops forces both sides to “out of
service.”
Wait 90 seconds (longer than Options Ping interval).
Repair circuit. Both sites return to “in service.”
Loss of circuit for brief moment (20 seconds; less than
the Options Ping interval). Active call should remain up
and voice path should reconnect.
Loss of circuit drops forces both sides to “out of
service.”
Active call should terminate gracefully on both sides.
Wait 300 seconds (longer than Options Ping interval).
Repair circuit. Both sites return to “in service.”
Power cycle PBX equipment with no active calls.
Trunk state should return to “in service” after reboot.
Pass
Power cycle PBX equipment with an active call. Call
should terminate gracefully. Trunk state should return
to “in service” after reboot.
Tele2 allows call from proper (authenticated) IP
Pass
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Page 6
80
Auth03b
authentication (Valid IP)
PBX to PSTN, source IP
authentication (Valid IP)
address of PBX.
Tele2 rejects call from invalid (non-authenticated) IP
address of PBX.
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Pass
Page 7
SIP Certification Testing Results – ShoreTel
ShoreTel-specific SIP trunk certification tests performed:
ID
1
Name
2.5
2
2.6
3
2.7
4
4.8
5
4.13
6
4.14
7
4.26
Title
Auto Attendant Menu
Description
Verify that inbound calls are properly terminated on the
ShoreTel Auto Attendant menu and that you can
transfer to the desired extension. (Multiple Digits)
Auto Attendant Menu “Dial Verify that inbound calls are properly terminated on the
by Name”
ShoreTel Auto Attendant menu and that you can
transfer to the desired extension using the “Dial by
Name” feature.
Auto
Attendant
Menu Verify that inbound calls are properly terminated on the
checking Voice Mail mailbox ShoreTel Auto Attendant menu and that you can
transfer to the Voice Mail Login Extension.
Conference – ad hoc
Verify successful ad hoc conference of three parties
(With SIP Media Proxy Ports enabled)
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.
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.
Call Recording
Verify that external calls can be recorded via the SIP
Trunk using ShoreTel Communicator
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Result
Pass
Pass
Pass
Pass
Pass
Pass
Pass
Page 8
ShoreTel Unsupported Features
ShoreTel only provides Technical Assistance for SIP trunk providers (ITSPs) that have been through
ShoreTel’s Innovation Network program. A current list of ShoreTel certified providers is located at
the following website:
http://www.shoretel.com/partners/tech_developers/ecosystem
Please refer to the ShoreTel Administration Guide, Chapter 18: “Session Initiation Protocol”, for more
information on supported and unsupported features when using SIP trunks.
The following are some feature limitations when using SIP trunks:
General Feature Limitations







Fax (and modem) redirection works only if the ITSP supports T.38. Fax redirection is not
supported on SIP trunks using G.711. Direct Inward Dialing (DDI/DID) to a specific fax
endpoint is still supported when using SIP trunks.
ShoreTel supports both jack-based and file-based Music on Hold (MoH) over SIP trunks.
When using jack-based MoH, it is required that the ShoreGear switch that hosts the SIP
trunks be configure with a jack-based music source. Jack-based MoH will not be streamed
over the network from another ShoreGear switch. File-based MoH does not have this
restriction.
The maximum number of music on hold (MoH) streams that a SIP-enabled switch can support
varies with the switch model. The range of such streams across all the voice switch models is
14–60.
Ad Hoc and MakeMe Conferencing: When a SIP trunk is involved in a 4- to 6-party
conference, Make Me conference ports will be used. For 3-party conference calls involving a
SIP trunk, the ShoreTel system will use SIP Media Proxy resources, if available. If no SIP
Media Proxy resources are available then 3 MakeMe Conference ports will be used.
The following features are supported by SIP only if the trunk has a SIP trunk profile with
Hairpinning enabled and the trunk is on a half-width switch with available SIP Media Proxy
resources: Silent Coach, Silent Monitor, Barge-In, Call Recording.
Extension Assignment with DTMF detection requires available SIP media proxy resources
and is supported without Hairpinning but is enhanced if Hairpinning is enabled and available.
See the ShoreTel documentation for additional details.
Silence detection on trunk-to-trunk transfers is not supported with SIP trunks.
There may be other feature limitations when using SIP Trunks. Please consult the ShoreTel
Administration Guide for further details.
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 9
ShoreTel Configuration
The configuration information below shows examples for configuring the ShoreTel UC system
settings required for connecting to the Tele2 SIP trunks. Configuration requirements will vary for
each environment but the information provided in these steps, along with the ShoreTel Planning and
Installation Guide and documentation provided by the SIP trunk provider should prove sufficient for
proper configuration and functionality. Of course, every design will vary and some environments will
require more planning, configuration and testing than others.
This section provides the general system settings and trunk configurations required for a ShoreTel
system to support SIP trunking.
Configuring Call Control Settings
The first settings to configure within ShoreTel Director are the Call Control Options. To configure
these settings for the ShoreTel system, log into ShoreTel Director and select “Administration | Call
Control | Options”.
Administration Call Control Options
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 10
In the Call Control Options window, under the “General” parameters, the “DTMF Payload Type (96 –
127)” defaults to a value of “102”. No modification is necessary to interoperate with Tele2.
Within the “SIP” parameters, set the following:
 The “Realm” setting can be left at the default of “ShoreTel”
 “Registration Expiration” can be left at the default of 60
 “Enable SIP Session Timer” should be checked
 “Session Interval” can be left at the default of 1800
 Refresher can be left at the default of “Caller (UAC)”
 “Always Use Port 5004 for RTP” parameters must be unchecked
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 11
Additional Details
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 a SIP trunk provider.
Verify that the “Enable SIP Session Timer” box is checked (enabled). The recommended setting
for “Session Interval” is between 1800 and 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 ShoreTel legacy parameter “Always Use Port 5004 for RTP” should be disabled by default,
if it’s enabled you will need to disable it. It is required for implementing SIP on the ShoreTel
system. For SIP configurations, dynamic User Datagram Protocol (UDP) must be used for RTP
Traffic. By disabling this parameter, both MGCP and SIP traffic will use dynamic UDP ports.
Changing this parameter requires a reboot of all ShoreTel devices and servers: IP Phones,
ShoreGear Switches, HQ and DVS Servers, Conference Bridges and Contact Center Servers.
By not performing a full system reboot, one-way audio can result.
The next settings to verify are the “Voice Encoding and Quality of Service” parameters, 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.
NOTE:
Trunk-to-Trunk transfer silence detection is not supported on SIP trunks.
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 12
Creating Codec Lists for Voice & Fax
Codec lists are used by the ShoreTel system to negotiate codec selection with the ITSP for each call.
Codec lists control what codecs are supported with an ITSP peer and in what prioritized order they
should be selected.
Tele2 prefers that PCM- A-law be used as the primary codec.
Within ShoreTel Director, navigate to “Administration | Call Control | Codec Lists.”
Create a New Codec List with the following parameters:
 Name:
“Tele2 - PCM Only”
 List Members: PCMA/8000
PCMU/8000
 List Order:
Select the “PCMA” codec and press “Move Up ^” to place it at the top of the
list
Save your changes.
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 13
We also need to make a new Fax Codec List.
NOTE:
The naming of a Fax Codec list must start with “Fax Codecs”.
Create a New Codec List with the following parameters:
 Name:
“Fax Codecs - Tele2”
 List Members: T.38
PCMA/8000
PCMU/8000
L16/8000
 List Order:
Use the “Move Up” and “Move Down” buttons to order the list as shown:
T.38, PCMA/8000, PCMU/8000, L16/8000
ShoreTel Site Settings
Next we need to address site settings. Modify these settings within ShoreTel Director by selecting
“Administration | Sites”.
Within the “Sites” screen, select the name of the site to be configured. The “Edit Site” screen
appears. The only changes required to the “Edit Site” screen is to the “Admission Control Bandwidth”
and the codec lists selected for “Intra-Site Calls”, “Inter-Site Calls” and Fax calls.
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 14
The Admission Control Bandwidth defines the WAN bandwidth available for use by voice calls to and
from this site. This is important as SIP trunk calls may be counted against the site bandwidth. See
the ShoreTel Planning and Installation Guide for more information.
Select the following parameters:
 Intra-Site Calls:
“Tele2 - PCM Only”
 Inter-Site Calls:
“Tele2 - PCM Only”
 Fax and Modem Calls: “Fax Codecs - Tele2”
Save your settings.
Configuring Switch Settings for SIP Trunks
The final general settings to configure are the ShoreGear switch settings. Modify these settings in
ShoreTel Director by selecting “Administration | Platform Hardware… | Voice Switches /Service
Appliances… | Primary”.
This will display the “Switches” screen. Click to edit the ShoreGear switch that will host the SIP
trunks. The “Edit ShoreGear Switch” screen will be displayed. Configure enough SIP Trunk
resources using either Built-in IP resources or resources from a physical port (e.g. Analog or BRI).
When using Built-in IP resources you can specify SIP trunk resources one-by-one
When assigning Analog ports or BRI channel resources you can choose to allocate either:
 5 SIP Trunk Resources, or
 1 SIP Trunk Resource with a SIP Media Proxy Resource
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 15
ShoreTel 13 introduces an additional option to the “Port Type” of half-width ShoreGear switches
called the “SIP Media Proxy”. “SIP Media Proxy” resources are used to ensure that SIP Trunks have
feature parity with PRI trunks. These features include support for RFC 2833 DTMF detection for
Office Anywhere/External Assignment and Simultaneous Ringing calls, three party conferencing
(without needing to configure “MakeMe” conference ports), Call Recording, Silent Monitoring, Silent
Coach, Barge-In, Whisper Page, Invites with no SDP and when there’s no common codec between
ITSP and the local extension, etc.
For further information on “SIP Media Proxy” please refer to Chapter 18 of the ShoreTel 13 System
Administration Guide.
If you are using older, full-width ShoreGear switches and you want to perform 3 (or more) party
conference calls with SIP trunks, make sure that you have enabled a minimum of four “MakeMe”
conference port resources. Conference resources are required with ShoreTel 13 on full-width
ShoreGear switches for 3-way conference calls to function. These resources may be on any switch
that has available ports and supports “MakeMe” conference resources at the site terminating the SIP
trunks. Alternatively, SIP Media Proxy resources on local half-width switches can be used for 3-party
calls involving SIP trunks.
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 16
Creating a SIP Profile
You need to create a new SIP Trunk Profile for use with Tele2’s SIP trunking
Within ShoreTel Director select “Administration | Trunks… | SIP Profiles”.
This action brings up the SIP Trunk Profiles page.
Click “New” and enter the following parameters:
 Name:
“Tele2 SIP Profile”
 User Agent:
“*.” (an asterisk and a period)
 Priority:
100
 Enable:
Checked
 Custom Parameters: OptionsPing=1
OptionsPeriod=30
DontFwdRefer=1
HistoryInfo=diversion
EnableP-AssertedIdentity=1
Hairpin=1
NOTE: These parameters are case sensitive.
Select “Save” at the top of the page.
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 17
These custom parameters are necessary to support specific features with Tele2 and ShoreTel SIP
Trunking:
Parameter
OptionsPing=1
OptionsPeriod=30
DontFwdRefer=1
HistoryInfo=diversion
EnablePAssertedIdentity=1
Hairpin=1
Description
Enables a heartbeat to be sent from the ShoreTel PBX to the ITSP
to keep the circuit active. This must be enabled or the trunks will be
marked as “out of service”.
A 30 Second interval is required to avoid the circuit from being taken
out of service. Setting the value above 30 seconds with Tele2 can
cause the trunks to be taken out of service by Tele2.
Inhibits the use of REFER for transfers and uses INVITEs instead.
Adds a Diversion header used to indicate the DID number of the
user on whose behalf the call was forwarded
Adds Caller-ID information in the P-Asserted-Identity header for
outbound calls
Enabled the use of ShoreTel SIP Media Proxy resources for
enhanced capabilities (PRI feature parity)
Creating the SIP Trunk Group
Next we will create the SIP Trunk Group.
In ShoreTel Director select “Administration | Trunks | Trunk Groups.”
Under “Add new trunk group at site” on the “Trunk Groups” screen, select the desired site and select
“SIP” as the trunk type. Click “Go”. The “Edit SIP Trunk Group” screen will appear.
Create a new Trunk Group with the following parameters:
 Name:
“Tele2 SIP Trunk Group”
 Enable SIP Info:
Unchecked
 Profile:
“Tele2 SIP Profile” (The profile created above)
 Digest Authentication: <None>
NOTE:
No registration is required by Tele2. Leave the Username and Password fields blank.
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 18












Inbound:
Digits from CO:
DID:
Outbound:
Access Code:
used 0>
Local Area Code:
Billing Telephone:
Local:
Long Distance:
National Mobile:
International:
Original Caller Info:
Caller ID not Blocked:
Dial in E.164 Format:
12
Checked
Checked
<Use an appropriate Truck Access Code for your deployment – we
<Use the appropriate LAC for your deployment – we used 20>
<Enter an appropriate BTN for your deployment>
Checked
Checked
Checked
Checked
Checked
Checked
Checked
Save your changes.
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 19
You will be asked whether you want all existing User Groups to be automatically configured with
permissions to use this new Trunk Group. You can click OK or click Cancel if you prefer to assign
this permission manually.
The parameter “Caller ID not blocked by default” determines if calls are sent out as <unknown> or
with Caller ID information (Caller ID). Tele2 requires that all calls go out with a valid and appropriate
CallerID number that matches the assigned DDI/DID range for each customer. DDI/DID information
is configured in the following section.
If you subscribe to a “Caller Line ID Restriction” service from Tele2 you can uncheck “Caller ID not
blocked by default”. Doing so will send all calls as “Anonymous” with the originating caller information
sent in the P-Assert ID field and marked as “ID” privacy.
DDI/DID Assignment
Tele2 requires that all outgoing calls be identified with proper CallerID information. Calls sent to the
PSTN that do not have the appropriate DDI/DID information in the SIP From: or P-Assert Identity
fields will be rejected by Tele2.
Earlier we configured Custom SIP Parameters related to CallerID in the Tele2 SIP profile. But you
must also create a range of DDI/DID numbers in the ShoreTel SIP Trunk Group and then assign
each user/entity an appropriate DDI/DID.
In ShoreTel Director, edit the SIP Trunk group and click the “Edit DID Range” button.
Add the proper DDI/DIDs that have been assigned to you by the ITSP.
Next, ensure that all users and all other ShoreTel entities that will be placing outbound calls have a
DDI/DID assigned to them.
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 20
Also ensure that all CESID values have a valid DID assigned as well. CESID numbers can be used
when calling emergency numbers. Plan you DDI/DID assignments carefully.
CESID numbers should be entered for all Sites, all IP Phone Address Maps and all switches. User
Group settings should be set according to how you desire CESID information to be sent on
emergency calls.
Creating Individual Trunks
Once the Trunk Group has been configured you can create the Individual SIP Trunks that will
connect to the ITSP. You will create one ShoreTel SIP Trunk for each active audio channel you have
ordered from the ITSP. If you have ordered 10 concurrent calls from the ITSP you will create 10
Individual SIP trunks in the ShoreTel system. If there are ten active calls, attempting to place an
additional outbound call will result in a fast “reorder tone” indicating that all trunks are busy. An
inbound caller will hear a busy tone from the ITSP indicating that no more trunks are available.
To create the individual trunks log into ShoreTel Director and select “Trunks | Individual Trunks.”
Select the proper site and select the “Tele2 SIP Trunk Group” from the drop down menus. Then
press “Go”.
Enter the following parameters:
 Name:
Tele2 SIP Trunk
 Switch:
<Select the ShoreGear switch which has the SIP Trunk Resources
assigned>
 IP Address:
<Enter the static IP address of the ITSPs SIP server/SBC/switch>
Note: This will be provided to you by the ITSP
 Number of Trunks: <Enter the number of trunks that you have ordered from your ITSP>
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 21
Save your changes.
This will create the number of Individual Trunks that you specified, all pointing at the IP address you
entered. The ShoreTel system will immediately attempt to put those trunks into service.
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 22
Custom Dial Strings for Tele2
Tele2 requires that several dialable numbers be translated by the ShoreTel system and forwarded
out the SIP trunks. These numbers include the following:
Dialed Number
112
1802
14020
116000
Translated Digits
+31 141 210112
+31 140 01802
+31 140 014020
+31 140 0116000
Usage
Emergency
Tele2 customer service
Amsterdam city
European child missing
112 is the default emergency number in Europe and is normally handled by the ShoreTel system and
sent out directly over an available trunk. Translating this to a full PSTN number requires custom dial
plan manipulation.
To support the additional special Tele2 number (1802, 14020 and 116000) we must add these
numbers as additional emergency numbers on the Site Edit page. These Emergency Numbers,
when dialed, need to be translated to the proper PSTN number and sent out the SIP trunk.
NOTE:
Because Tele2 requires these numbers to be dialable without a Trunk Access Code and requires
that they be translated to full PSTN numbers the ShoreTel system must be configured so that the
leading digit of “1” is reserved (“not used”). Therefore you cannot have any extensions, trunk
access codes or other ShoreTel entities that begin with “1”.
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 23
Re-routing Emergency Numbers
With normal PSTN trunks, an emergency number is sent out to the PSTN directly.
Tele2 prefers to have emergency and customer service numbers translated to a fully qualified PSTN
number. In some case (+31 140 01802) the number is not a valid number according to national
standards but is captured and used internally by Tele2.
Four emergency numbers must be configured this way in order to pass the certification tests provided
by Tele2.
There are four steps in achieving this configuration:
1. Add the Emergency numbers to the Site page in ShoreTel Director
2. Ensure that the leading digit of “1” is manually set to “Not Used”
3. Obtain and sign a ShoreTel waiver form in which you agree to have your emergency
numbers altered from the default configuration
4. Engage the ShoreTel Implementation Service team to assess your configuration, design and
implement a custom dial plan to translate the numbers and then test the solution for proper
functionality
Please contact your ShoreTel sales representative for additional details to arrange for steps 3 and 4
during your deployment process.
Steps 1 and 2 are discussed below.
Step 1: To define the emergency numbers perform the following in ShoreTel Director:
 Navigate to “Administration | Sites”
 Edit your Site
 Scroll to the bottom
 Under “Emergency Number List” enter your first number (“112”)
 Click the “Add More” button
 Enter the second emergency number (“14020”)
 Click the “Add More” button
 Enter the final emergency number (“116000”)
 Save your changes
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 24
Step 2: To reserve the leading digit of “1” do the following in ShoreTel Director:
 Navigate to “Administration | System Directory”
 Click the “Ext” column heading to sort by Extension
 Examine your extensions. If any extension or entity start with “1xxx” you must edit and
change them to start with a different leading digit

Confirm that nothing is using a leading digit of “1” (this includes Trunk Access Codes, etc.)
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 25



Navigate to “Administration | System Parameters | Dialing Plan”
Change Digit “1” to “Not Used”. If Digit “1” cannot be changed then some entity or Trunk
Access Code is still configured to use “1” as a leading digit
Save your changes
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 26
Customizing CallerID on Forwarded Calls
One last parameter needs to be configured to adjust the presentation of CallerID information when a
user forwards incoming PSTN calls to an external PSTN number.
Normally, the ShoreTel system would attempt to place the original Calling Party’s CallerID
information on the forwarded outgoing call. In the case of Tele2 trunks the original caller ID is being
replaced with “Anonymous”.
As a security measure, Tele2 prefers that a valid number, within your assigned DDI/DID range, be
inserted instead of “Anonymous”. This allows them to verify and validate that the call is not
attempting to present a false or invalid CallerID.
To enable this feature you must enter a parameter as a Custom Dial Plan setting.
Perform the following:
 If you are logged into ShoreTel Director, logout (but leave the browser open)
 Press and hold the CTRL + SHIFT keys on your keyboard and click on the word “Username:”
 This will enable “*** Support Entry ***”





Enter your normal user name and password to log in as an Administrator
Navigate to “Administrator | Trunks | Trunk Groups”
Select and edit the “Tele2 SIP Trunk Group”
Scroll to the bottom of the page
You will see a new set of parameters allowing you to view and edit the dial plan for this trunk
group


Click the “Edit” button
Enter the following: ;1S (a semicolon, the number one and a capital ‘S’)
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 27

Save your changes
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 28
Conclusion
Tele2 provides a robust SIP trunking solution and, with proper configuration, the ShoreTel system
supports the Tele2 SIP trunks extremely well.
Document & Software Copyrights
Copyright © 2013 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.
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 are
trademarks of ShoreTel, Inc. in the United States and/or other countries. All other copyrights and
trademarks herein are the property of their respective owners. .
Disclaimer
ShoreTel tests and validates the interoperability of the Member's solution with ShoreTel's published
software interfaces. ShoreTel does not test, nor vouch for the Member's development and/or quality
assurance process, nor the overall feature functionality of the Member's solution(s). ShoreTel does
not test the Member's solution under load or assess the scalability of the Member's solution. It is the
responsibility of the Member to ensure their solution is current with ShoreTel's published interfaces.
The ShoreTel Technical Support organization will provide Customers with support of ShoreTel's
published software interfaces. This does not imply any support for the Member's solution directly.
Customers or reseller partners will need to work directly with the Member to obtain support for their
solution.
Company Information
ShoreTel, Inc.
960 Stewart Drive
Sunnyvale, California 94085 USA
+1.408.331.3300
+1.408.331.3333 fax
960 Stewart Drive Sunnyvale, CA 94085 USA
Phone +1.408.331.3300 | +1.800.425.9385 | Fax +1.408.331.3333
www.ShoreTel.com
Page 29