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Technical
Configuration Notes
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Mitel Technical Configuration Notes Configure the Mitel 5000 CP for use with the Polycom
SoundStation IP 7000
January 7, 2011, 11-4940-00158
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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
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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:
Variant
Software Version
Mitel
5000 CP
4.0 release 26
Polycom
SoundStation IP 7000
3.3.1.0769
Mitel
7.2.1.5.0.41
Mitel
5330 IP Sets
1.6.2.3
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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
DTMF Signal
Call Hold
Call Transfer
Call Forward
Conference
Redial
MWI
T.38 Fax
Fax Messages
Not Applicable
Video
Video Capabilities
Not Applicable
- No issues found
Issues
- 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
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
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Network Topology
This diagram shows how the testing network is configured for reference.
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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.
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).
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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
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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.
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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.
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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)
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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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