11-4940-00158_-_polycom_ip_7000

MITEL SIP CoE

Technical Configuration Notes

Configure the Mitel 5000 CP for use with the Polycom SoundStation IP 7000

SIP CoE 11-4940-00158

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NOTICE

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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

, Trademark of Mitel Networks Corporation Copyright 2011, Mitel Networks Corporation

All rights reserved

Table of Contents

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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

11-4940-00158 Polycom SoundStation IP 7000

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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 Date Reason

1 January 2011 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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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 Issues

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 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 Problem Description

Registration failure 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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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.

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


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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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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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The Polycom SoundStation IP 7000 was configured as displayed below.

Figure 3 Polycom SoundStation IP 7000 Configuration


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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

Figure 4 Password Configuration


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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 --> --> 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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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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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 near-end NAT)

Figure 9 NAT Settings


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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 un-registers.

This folder allows you to configure the following settings that are required for registration per-

Polycom 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\). When you create a SIP peer without using a template, by default the new SIP peer is added to Call Configuration 1 .

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 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

11-4940-00158_-_polycom_ip_7000

Documents

trademarks mitel

cp configuration notes

sip phone configuration

polycom soundstation

sip phone groups

cp programming

software license sip

interop history