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MITEL SIP CoE

Technical
Configuration Notes

Configure the Mitel 5000 CP for use


with the Polycom SoundStation IP
7000
SIP CoE 11-4940-00158

NOTICE
The information contained in this document is believed to be accurate in all respects but

is not warranted by Mitel Networks Corporation (MITEL ). The information is subject to


change without notice and should not be construed in any way as a commitment by Mitel
or any of its affiliates or subsidiaries. Mitel and its affiliates and subsidiaries assume no
responsibility for any errors or omissions in this document. Revisions of this document or
new editions of it may be issued to incorporate such changes.
No part of this document can be reproduced or transmitted in any form or by any means electronic or mechanical - for any purpose without written permission from Mitel Networks
Corporation.

TRADEMARKS
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Other product names mentioned in this document may be trademarks of their respective
companies and are hereby acknowledged.

Mitel Technical Configuration Notes Configure the Mitel 5000 CP for use with the Polycom
SoundStation IP 7000
January 7, 2011, 11-4940-00158

, Trademark of Mitel Networks Corporation


Copyright 2011, Mitel Networks Corporation
All rights reserved

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Table of Contents

OVERVIEW ............................................................................................................... 1
Interop History....................................................................................................................1
Interop Status ....................................................................................................................1
Software & Hardware Setup...............................................................................................1
Tested Features.................................................................................................................2
Device Limitations..............................................................................................................3
Network Topology ..............................................................................................................4

CONFIGURATION NOTES ....................................................................................... 5


5000 CP Configuration Notes..........................................................................................5
Network Requirements.................................................................................................................... 5
Assumptions for the 5000 CP Programming................................................................................... 5
Software License SIP Licensing................................................................................................... 6
5000 SIP Phone Configuration........................................................................................................ 7
SIP Phone Groups ........................................................................................................................ 10

POLYCOM SOUNDSTATION IP 7000 CONFIGURATION NOTES ....................... 18


Digitmap Assignment .......................................................................................................23

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Polycom SoundStation IP 7000

Overview
This document provides a reference to Mitel Authorized Solutions providers for configuring
the Mitel 5000ICP to host the Polycom SoundStation IP 7000. The different devices can be
configured in various configurations depending on your VoIP solution. This document covers
a basic setup with required option setup.

Interop History
Version
1

Date
January 2011

Reason
Initial Interop with Mitel 5000 CP 4.0 Release 26 and the Polycom
SoundStation IP 7000

Interop Status
The Interop of the Polycom SoundStation IP 7000 has been given a Certification status. This
device will be included in the SIP CoE Reference Guide. The status the Polycom
SoundStation IP 7000 achieved is:

The most common certification which means the device/service has


been tested and/or validated by the Mitel SIP CoE team. Product
support will provide all necessary support related to the interop, but
issues unique or specific to the 3rd party will be referred to the 3rd party
as appropriate.

Software & Hardware Setup


This was the test setup to generate a basic SIP call between the Polycom SoundStation IP
7000 SIP device and the 5000 CP.
Manufacturer

Variant

Software Version

Mitel

5000 CP

4.0 release 26

Polycom

SoundStation IP 7000

3.3.1.0769

Mitel

5330 SIP Sets

7.2.1.5.0.41

Mitel

5330 IP Sets

1.6.2.3

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Polycom SoundStation IP 7000

Tested Features
This is an overview of the features tested during the Interop test cycle and not a detailed view
of the test cases. Please see the SIP Line Side Interoperability Test Pans for detailed test
cases.

Feature

Feature Description

Basic Call

Making and receiving a call

DTMF Signal

Sending DTMF after call setup (i.e. mailbox password)

Call Hold

Putting a call on hold

Call Transfer

Transferring a call to another destination

Call Forward

Forwarding a call to another destination

Conference

Conferencing multiple calls together

Redial

Last Number Redial

MWI

Message Waiting Indication

T.38 Fax

Fax Messages

Not Applicable

Video

Video Capabilities

Not Applicable

- No issues found

Issues

- Issues found, cannot recommend to use

- Issues found

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Device Limitations
This is a list of problems or not supported features when the Polycom SoundStation IP 7000
SIP device is connected to the Mitel 5000.
Feature
Registration failure

Problem Description
Register DUT with 5000 CP with an invalid password, 5000 challenged
with 401, DUT sent re-register with credentials, 5000 replied with 401
again instead of 403, then DUT sent re-register again with credentials
Recommendation: Contact Mitel for updates on registration process.

Registration using
Hostname

The 5000 CP currently does not support registration of SIP endpoints by


using the hostname of the 5000.
Recommendation: Use the IP address of the 5000 for registration.

Unsupervised call transfer

SIP call transfer (unsupervised) does not get ringback tone,


Phone A called phone B and phone B unsupervised transfered
the call to phone C, prior to the call being answered by phone C,
phone A did not get ringback tone. After phone C answered the
call, two way audio confirmed.
It only happened when both phone A and phone B were sip
phones. If either phone A or phone B was an IP phone or TDM
phone, then no such issue. Phone C can be any type of phones.
Recommendation: Contact Mitel for updates on this feature.

Dynamic Extensions

Dynamic Extensions was not fully supported for SIP phones until v5.0
Recommendation: Contact Mitel for updates on this feature

Forwarding Paths

System based forwarding Paths is not available for SIP phones, though
it works for Minet IP phones
Recommendation: Contact Mitel for updates on this feature

MWI Subscribe

5000 does not support Subscribe method in 4.0, it will be supported in


v5.0
Recommendation: Contact Mitel for updates on this feature

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Polycom SoundStation IP 7000

Network Topology
This diagram shows how the testing network is configured for reference.

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Polycom SoundStation IP 7000

Configuration Notes
This section is a description of how the SIP Interop was configured. These notes should give
a guideline how a device can be configured in a customer environment and how the Polycom
SoundStation IP 7000 was configured in our test environment.
For more detailed information on the programming of the Mitel 5000 CP please refer to the
Mitel 5000 CP Features and Programming Guide.
Disclaimer: Although Mitel has attempted to setup the interop testing facility as closely
as possible to a customer premise environment, implementation setup could be
different onsite. YOU MUST EXERCISE YOUR OWN DUE DILIGENCE IN REVIEWING,
planning, implementing, and testing a customer configuration.

5000 CP Configuration Notes


The following steps show how to program a 5000 CP to connect with the Polycom
SoundStation IP 7000 Phone.

Network Requirements

There must be adequate bandwidth to support the voice over IP. As a guide, the Ethernet
bandwidth is approx 85 Kb/s per G.711 voice session and 29 Kb/s per G.729 voice
session (assumes 20ms packetization). As an example, for 20 simultaneous SIP
sessions, the Ethernet bandwidth consumption will be approx 1.7 Mb/s for G.711 and
0.6Mb/s. Almost all Enterprise LAN networks can support this level of traffic without any
special engineering. Please refer to the 5000 Engineering guidelines for further
information.

For high quality voice, the network connectivity must support a voice-quality grade of
service (packet loss <1%, jitter < 30ms, one-way delay < 80ms).

Assumptions for the 5000 CP Programming

The SIP signaling connection uses UDP on Port 5060.

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Polycom SoundStation IP 7000

Software License SIP Licensing


Ensure that the 5000 CP is equipped with enough Category F Phones licenses for the
connection of SIP end points. This can be verified within the Software License Feature
section form.

Figure 1 Software License

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Polycom SoundStation IP 7000

5000 SIP Phone Configuration


To create an extension for the Polycom SoundStation IP 7000:
1. Select System Devices and Feature Codes Phones.
2. Right-click anywhere in the right pane, and then select Create SIP Phone. The Create
SIP Phone Extension dialog box appears.
3. Select a starting extension for the phones and the number of extensions.
4. Click OK. The system creates a new SIP Phone Group for each of the Polycom
SoundStation IP 7000s.
The SIP Phone groups are created in a stand-alone configuration by default. The associated
SIP Phone Group is displayed in System\Devices and Feature Codes\SIP Peers\SIP Phone
Groups
.

Figure 2 Create SIP Extension

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Polycom SoundStation IP 7000

The Polycom SoundStation IP 7000 was configured as displayed below.

Figure 3 Polycom SoundStation IP 7000 Configuration

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Polycom SoundStation IP 7000

The Password field is for the SIP authentication password and the username is the DN. All
other field names should be programmed according to the site requirements or left at default

Figure 4 Password Configuration

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Polycom SoundStation IP 7000

SIP Phone Groups


The Polycom SoundStation IP 7000 can register with the 5000 CP and act as local
extensions in the system. To support this feature, DB Programming uses SIP Phone
Groups. A SIP Phone Group contains a common set of properties for registration that can be
shared with either a stand-alone SIP Phone Group or multiple SIP Phone Group.

Figure 5 SIP Phone Groups


Authentication
The Polycom SoundStation IP 7000 SIP Phone group was configured to use In Bound
Authentication:
Enable In-Bound Authentication: If the Enable In-Bound Authentication flag is enabled
For a SIP peer, incoming calls and SIP requests from the SIP peer are authenticated by the
5000 CP.
.

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Figure 6 SIP Phone Groups Authentication Parameters

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Inbound Authentication Realm


The Realm for each SIP Peer is defined at the following location:
System --> Devices and Feature Codes --> SIP Peers --> SIP
(Trunk/Phone) Groups --> <xxxx> --> Configuration --> Authentication -->
In-bound Authentication Realm.
However, if this value is empty then system looks for a value defined
for all SIP Peers at following location:
System --> Devices and Feature Codes --> SIP Peers -->General
Configuration --> In-bound Authentication Realm.
If this value is also not set, then a default value of
'Mitel-5000-ICP' is used.
Note that the above configuration settings are available only in OLM
(On-Line Monitor) mode of Database Programming.
Also, this realm is used only for authenticating incoming SIP requests, if inbound authentication is
enabled.

Figure 7 SIP Phone Groups Authentication-In-bound Realm

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Polycom SoundStation IP 7000

MWI
The Message Waiting Indication (MWI) field determines whether the system accepts the MWI
From the SIP peer. Verify that the Accept MWI option is set to Yes. To have the system
ignore MWI from the SIP peer, change the setting to No. It is set to Yes by default.

Figure 8 MWI Configuration

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Polycom SoundStation IP 7000

For SIP Phone Groups


There are two choices:
Native: Used for internal Polycom SoundStation IP 7000s (those that do not pass through
near-end NAT).
NAT: Used for external Polycom SoundStation IP 7000s (those that do pass through nearend NAT)

Figure 9 NAT Settings

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Polycom SoundStation IP 7000

Registrations
You can register Polycom SoundStation IP 7000 with the 5000 CP dynamically
For dynamic registration, the status of a Polycom SoundStation IP 7000 is determined by
the existence of an active registration in the system for that Polycom SoundStation IP 7000.
When a Polycom SoundStation IP 7000 registers with the system, its status becomes Idle
(online) as long as there is a valid Polycom SoundStation IP 7000(Category F Phones)
license available and becomes Offline when the registration expires or SIP phone unregisters.
This folder allows you to configure the following settings that are required for registration perPolycom SoundStation IP 7000 Group basis:
Address of Record: Indicates the Address of Record that the SIP peer uses to register
with the 5000 CP. This field is for read-only.
Registration URI: Indicates the SIP URI representing the Contact address in the SIP
REGISTER request from the SIP peer that created this dynamic binding. This field is for
read-only.
Registration Call ID: Indicates the SIP Call ID of the SIP REGISTER request received
from the SIP peer that created this dynamic binding. This field is for read-only.
Registration Cseq Number: Indicates the SIP Cseq number of the SIP REGISTER
request received from the SIP peer that created this dynamic binding. This field is for
readonly.
Registration Update Time: Indicates the timestamp when the SIP REGISTER request
was received from the SIP peer and updated the dynamic binding. This field is for readonly.
Registration Expire Time: Indicates the time in seconds to expire this registration since it
was last updated. This field is for read-only.

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Figure 10 Registrations

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Call Configuration
Clicking Call Configuration takes you to the Call Configuration folder (System\IP-Related
Information\Call Configurations\<call configuration number>). When you create a SIP peer
without using a template, by default the new SIP peer is added to Call Configuration 1
<Local>.
The following diagram shows call configuration was used with the Polycom SoundStation IP
7000.

Figure 11 Call Configuration

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Polycom SoundStation IP 7000

Polycom SoundStation IP 7000 Configuration Notes


The following steps show how to program the Polycom SoundStation IP 7000 to interconnect with the
5000 CP.

1. Connect the Polycom SoundPoint IP 7000 SIP phone to the network and mains power
supply. The phone will boot up, and providing you have DHCP installed on your network, will
obtain an IP address. To find the IP address, using the phone keypad, go to menu -> status > network -> tcp-ip parameters and view its IP address.
2. Now you can access the phones web configuration page by going to http://x.x.x.x where
x.x.x.x is the IP address of the phone from step 1.
3. The factory default username and password for accessing web configuration on Polycom
phones is Polycom and 456 (make sure Polycom starts with a capital P).
4. In the General Section under General Configuration Parameters, we have only modified
the Audio Processing section by setting the G.711u codec to first, the G.711A to second
and the G729AB to third.

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5. In the Network Section under Network Configuration Parameters, we have not modified
any settings.

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6. In the SIP Section under SIP Configuration Parameters, we have modified the Servers
section as follows:
a. Set the Address and Port under Server 1 subsection to the same values as line setting.
b. Set the Expires field to a suitable value(>=300s). A good setting for general use could be
3600 seconds (1 hour).

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7. In the SIP Section under SIP Configuration Parameters, we have modified the Local
Settings section just by setting the Local SIP port to 5060

8. In the LINES -> LINE Parameters -> Identification. Display Name is for display purposes
only with no direct effect on the functionality of the phone. The users first and last name for
this extension is a reasonable choice here.
9. Set Address with the extension number that you set in 5000 CP.
10. Set Auth user ID and Auth password to the same values you have configured in 5000 CP
as User ID and Login password.
11. Set Label to what you want to appear on the screen of your phone. The Extension number
may be a good choice here.
12. Set the Address under Server 1 subsection field to the IP address of the server on which
the Mitel 5000 CP is installed.
13. Set the Port to 5060. Please note that if the SIP port has been changed from the default on
the 5000 CP, this value must be changed accordingly.

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14. Set the Expires to a suitable value(>=300s). A good setting for general use could be 3600
seconds (1 hour)

15. Submit the changes and start to place a call.

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Digitmap Assignment
Modifying the Digitmap for ARS Access codes:
On the Polycom web portal, under SIP, choose local settings. In the digitmap field the default
value is set to:
[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT
This translates as follows:
[2-9]11...........................211, 311, 411... 911
0T...................................operator (dial zero after timeout)
011xxx.T.........................International Calling (dial after timeout)
[0-1][2-9]xxxxxxxxx........0(zero) + and 1 + dialing North America
[2-9]xxxxxxxxx................10 digit local calling
[2-9]xxxT........................4 digit internal dialing (after timeout)
The example below would include a Digit 9 for ARS Access:
9[2-9]11|90T|9011xxx.T|9[0-1][2-9]xxxxxxxxx|9[2-9]xxxxxxxxx|[1-8]xxx|0T
9[2-9]11..........................9211, 9311, 9411... 9911
90T.................................External operator (dial zero after
timeout)
9011xxx.T.......................International Calling (dial after timeout)
9[0-1][2-9]xxxxxxxxx........0(zero) + and 1 + dialing North America
9[2-9]xxxxxxxxx..............10 digit local calling
[1-8]xxx..........................4 digit internal dialing
0T..................................Internal Zero - Switch Board
Optionally other dialing strings can be included for Feature access codes
or Speeddials
Example below includes *8 to dial voicemail and *1xxx to dial system
speeddials that start with *1
9[2-9]11|90T|9011xxx.T|9[0-1][2-9]xxxxxxxxx|9[2-9]xxxxxxxxx|[1-8]xxx|0T|*8|*1xxx

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