HowTo configure Polycom SoundPoint IP phones with sipX
From SIPfoundry sipXecs IP PBX, The Open Source SIP PBX for Linux - Calivia
Introduction
| Important note: Firmware limitations of Polycom phones: Even the most recent firmware (2.1.x) has some severe limitations that affect certain features. Polycom phones do not support standards compliant mechanisms for Music on Hold (MoH). Polycom phones have a bug in the handling of dialog events and therefore do not properly cooperate with the sipX presence server for BLF. We hope Polycom will address these limitations. |
sipX, The open source SIP PBX for Linux supports the full lineup of Polycom's SIP phones. Polycom�s standards-based desktop IP telephones provide lots of features, excellent sound quality and offer hands-free full-duplex speakerphone technology (one-way, monitor speaker in SoundPoint IP 300/301). Supported models include
- SoundPoint IP-300 / IP-301
- SoundPoint IP-430
- SoundPoint IP-500 / IP-501, IP-550
- SoundPoint IP-600 / IP-601
- Soundpoint IP-650
- SoundPoint Attendant Console
- Soundstation IP-4000
The new low-end models IP-320 and IP-330 will be added in the 3.10 release.
Useful Links
All the SoundPoint phones (30x, 430, 50x, 60x, 650, 4000) are supported through the The sipXecs IP PBX Configuration Server. That means that the phones are configured to use sipX as their boot server, from which to optain configuration profiles at startup. These configuration profiles are automatically generated by sipX ConfigServer, where administrators use the sipX Web UI to define all necessary parameters.
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Model IP601 and the Attendant Console: The Polycom Attendant Console is a great addition to the IP601 and IP650 phones. With the attendant console attached, the Polycom IP601 / IP650 phones are able to support up to 12 lines. In conjunction with the corporate directory management feature it allows presentation of speed dial keys. Line state of other phones can be monitored using a feature called Busy Lamb Field (BLF). The phone subscribes to the state of a line on another phone and displays that state using the indicator light next to the line key. sipX implements a centralized presence server that with Polycom's firmware release 2.x supports BLF. We are also working on building support for true bridged line appearances (BLA) based on presence indication. This will require release 2.x SIP firmware for the phone, as well as some additional functionality in the sipX server. |
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New Model IP650 with HD Voice: sipX delivers better voice quality because media streams are always routed peer-to-peer along the shortest path and not through the PBX controller as is the case with most other systems. This allows full support for higher bandwidth codecs such as Polycom's HD Voice (using G.722 wide-band codec) without imposing any restrictions on how many calls can be active at any given time. Polycom HD Voice delivers over twice the clarity of ordinary phone calls for life-like, vibrant conversations. The difference is so astounding, you will never want to go back to regular phone calls. sipX (release 3.6) fully supports the IP650 including HD Voice. | 
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Models IP301 and IP501: The main difference between the IP301/501 and the old IP300/500 is additional memory built into the phones to accomodate the growing size of the SIP firmware image. The IP600 already has this extra memory built in. This means that the old models IP300/500 will not be upgadable to new firmware after SIP 1.4.1. Some of the new features, such as HTTP, HTTPS, and FTPS, will therefore not be supported.
Note: The Polycom phones come with an AC power adaptor included. However, the IP30x and IP50x models do not support Power Over Ethernet (PoE). An additional external special cable is required to make the phone support 802.3af and the Cisco PoE standard. Apparently, the cable contains the PoE hardware.
Steps Required to Commission a New Phone
Network Configuration before you start
In order for the phone to recognize the sipX server as its boot server at startup, your DHCP server needs to provide additional necessary information to the phone as it obtains an IP address. By default the phone is configured to look for the DHCP option Boot Server Host Name to obtain the IP address or host name of the (T)FTP server that is configured to provide the configuration information.
Using a Windows DHCP server: The Boot Server Host Name parameter is the variable with the number 66 (Option 66) in the DHCP Scope Options settings.
Using a Linux DHCP server: Add the following line to your DHCP server config file:
option tftp-server-name "your sipxpbx hostname";
During installation of the sipX server, both an FTP and TFTP service option can be automatically configured and enabled, where both can coexist at the same time. The phone by default tries to contact an FTP server and we therefore recommend this option for use with Polycom phones.
For more information on using FTP w/polycom see Setting up Polycom to use FTP page.
Note: If it is not possible to change the configuration of your local DHCP server, the phone can be configured manually with a (T)FTP server IP address or host name (see below)
Set up the phone using the sipX Configuration Server
Once the phone is able to find its boot server at startup, all remaining configuration is done automatically and remotely using the sipX configuration server and Web UI. The tasks required to complete the installation are as follows:
- Create at least one user who will be using the phone
- Create the phone as a device known to sipX
- Assign one or several users to the phone (create lines)
- Generate the profile for the phone and reboot the phone
Note: If the phone is unable to find its boot server, it displays a clearly visible error message at the beginning of the boot sequence. If this message does not come up the phone successfully communicated with its boot server.
1. Create a user
- Login to the sipX UI as superadmin
- It is recommended that you add different User Groups as Web UI and Call Handling Permissions can be defined either for groups or individual users. Groups can consist of users with different phones.
- Click on Users from the main navigation, then click the Add User button to create a new user:
- Create a new user by entering the requested parameters. The extension number is populated based on the settings in User Extension Pool; however, it can be changed. Note that the PIN is only the password the user needs to login to the voicemail portal. The SIP password the phone will use for registration is automatically generated and assigned. It can be viewed / changed by clicking on Show Advanced Settings. (See here for more information).
5. Edit permissions by going to Group Settings for the group the user belongs to. If the user should be able to record system prompts using the phone, enable it. Also, make sure you set correct permissions for dialing rules. Alternatively, permissions can be set for an individual user.
2. Create a Device for the new phone
- Click on Phones in the main navigation, then Add Phone. Enter each phone's MAC address (lower case letters only) as the serial number, choose the appropriate phone model, and place it into the desired group.
2. Click on your new phone entry, listed by its serial number, then go to Lines.
Note: The outbound proxy field is intentionally left empty. The line will register with the server specified as Registration Server for each line. This allows several lines to register with different proxy server.
3. Assign user(s) to the phone
- Click on Add Line. You can search for a specific user or just click Search with empty fields, which will list all users. Select a user and press Select. You have now attached that user to the device. Dependent on the phone model, several lines can be attached to a device.
4. Generate the profile
1. Click the checkbox on the left side of the entry and select the Send Profiles button. This now creates the configuration profile for the phone with default parameters and stores it in the TFTP root directory of the sipX server. During the previous steps we have used default parameters for both the user we created as well as the device. These default parameters should allow the phone to properly register with sipX when it boots next time. Once registration is accomplished, user and device parameters can be changed to your specific requirements.
Note: You can verify that the profile was generated successfully by clicking on Job Status under Diagnostics in the main navigation menu. You will find an entry for the newly generated profile.
2. You now need to restart your Polycom phone manually by unplugging the power cord (or press "+", "-", "Messages" and "Mute" all at the same time). As the phone comes back up it should now successfully register with the sipX proxy. This can be verified by clicking on Registrations under Diagnostics in the main navigation menu. An entry should be shown for the phone. Once the phone registered successfully with ConfigServer it can then be restarted from the Web UI remotely as you make further changes to its configuration.
5. Phone Internal Dial Plan Settings
The phone internal dial plan decides when sufficient digits have been entered for the phone to start establishing the session. This phone internal dial plan string has to be inline with the dial plan defined for the sipX server and some of the choices offered. Please refer to the Digit Maps page for further details.
6. Firmware versions
sipXconfig supports Polycom firmware releases 1.6.x and 2.x. By default newly created Polycom devices (or imported Polycom devices) are presumed to have 1.6 firmware installed. When adding or editing the phone you can tell sipXconfig which firmware version it uses (please not that 2.0 really means any 2.x firmare). Changing the firmware version in sipXconfig does not upgrade or downgrade your phone: it just changes the format of configuration files generated by sipXconfig. You can use Device Files feature to install new version of the firmware and upgrade your phone.
If you have upgraded the phones and want to tell sipXconfig to generate 2.x compatible files execute:
sipxconfig.sh --database polycom-2.0
Similarly you can switch all the phones to 1.6
sipxconfig.sh --database polycom-1.6
There is a property in sipxconfig.properties.in called
polycom.defaultVersionId
if you set it to 2.0 (it really means 2.x) and restart sipXconfig all newly added or imported Polycom devices will by default have 2.x compatible files generated.
7. Additional Information
Enable Additional Features
Voicemail Notification (Message Waiting Indication - MWI)
Using default configuration settings, your Polycom phone will subscribe to the sipX Media Server for voicemail services. To verify or change settings go to the configuration screen for the line in question and click on Messaging.
Set Date / Time using a NTP Time Server
Default settings should be sufficient to let your phone synchronize its date and time settings. To verify / change do the following:
Open the main configuration screen for the Polycom phone in question go to the SNTP menu Enter pool.ntp.org into the address field (2nd from the top) Adjust the field gmtOffset Save configuration by clicking on Apply or OK Select the phone using the check box on the left and hit Send profile The phone should now reboot and then display the correct date / time
Ring Tones and Volume
Additional features can be configured using the phone top interface. Such configurations are stored as user specific configurations or User Settings and uploaded to the configuration server.
Ring Tones: Press Menu and go to Settings, press select. Go to Ring Type and select the line and then scroll down to select the desired ring tone.
Volume: Volume can be adjusted by simply pressing the volume keys below the key pad at any time.
Note: Custom Ring Tone files can be made available using the TFTP server.
Emergency Numbers and E911 Call Routing
There are three different ways of how to setup emergency calling in sipX. They are not mutually exclusive but can all be configured and used simultaneously. The list below indicates the recommended preference for the three available options:
- If the phone is capable of routing emergency calls directly to a PSTN gateway bypassing the sipX server, then this is the most reliable and therefore preferred solution. Polycom phones offer this option (see below)
- Activate the Emergency Dialing rule in the sipX default dial plan. This will establish a routing rule for the E911 emergency number in combination with the normal prefix for an outside line (e.g. 9)
- Use the Emergency Routing screen from the main navigation in sipX ConfigServer and define an emergency gateway to be used for E911 emergency calls. When using option 2 above the default gateway is used for all emergency calls. The Emergency Routing option allows you to define two different gateways dependent on the calling number (How to configure caller sensitive forwarding for emergency calls).
Polycom phones support direct routing of emergency calls to a defined PSTN gateway. This provides the highest possible reliability as the call can be completed even if the sipX server is currently not available. It is therefore recommended to configure this option as follows using the sipX ConfigServer:
Select a phone and go to the Polycom phone configuration screen and click on Lines Chose the line for which you want to enable direct E911 emergency calling Click on Dial Plan and scroll down to Emergency Gateway Into the address field, enter the IP address or FQHN of the gateway to which you want to route emergency calls Enter a port; default is 5060 Into the Emergency Numbers field enter the numbers or URLs a user would dial to reach emergency services Click OK to save Repeat this for every line configured on the phone Remember to select Send Profiles when done, which will cause the phone to reboot
Boss - Secretary Relationship
Several features are useful to establish a boss - secretary relationship. Some of these features require additional functionality in the SIP server.
1. Call Forwarding:
The Polycom phones support call forwarding to a defined contact on busy, on no answer, and on do not disturb. Using the sipX ConfigServer go to Phones, <a specific phone>, Lines, <a specific line>, Diversion to configure this feature.
2. Bridged Line Appearance (BLA):
Bridged line appearances are generally used in larger PBX environments. The most common use case is boss-secretary relationships. This feature is very common in professional businesses like law firms or accounting firms. Generally there is an assistant that may be managing calls for 1-3 partners. The assistant would have line/extension appearances of each line that the boss' phone has.
If two phones are configured with a bridged line appearance, an incoming call to that number will cause both phones to ring. The call can be answered on either phone but not both. When the call is placed on hold, it will be accessible from both phones. If a bridged line is in use on one of the phones, it cannot be used on the other one. The bridged line icon on the phone changes to a moving icon and the line LED turns red (IP600) when the line is in use remotely.
While the Polycom phones firmware 2.x support this feature, it is not yet fully implemented in the sipX server. It is possible today to assign the same line to two different phones. This causes both phones to ring and the call can be picked up on either phone. It is further possible to see whether the remote line is in use using the BLF feature. However, the line can be used on either phone simultaneously to initiate calls and putting the call on hold on one phone does not allow retrieving it on the other (a transfer is required). Therefore there is no real "shared line" experience yet.
More information on how to implement bridged line appearances in SIP can be found in this IETF draft.
3. Call Park & Retrieve:
Call Park & Retrieve is often used in combination with bridged line appearances in a boss-secretary relationship. If the boss is in a call and the secretary picks up a new incoming call, he / she will need a possibility to transfer the call back to the boss once he / she is off the phone. This can either be done by assigning an additional private line to the boss' phone to which the secretary would transfer the call, or it can be done using the call park & retrieve feature. In this case the secretary would transfer the call to one of the park extensions, which effectively puts the call on hold. The boss can then pick it up from there usually by pressing a dedicated key.
4. Presence Indication:
The Polycom phones are able to indicate presence within a pool of phones. Each phone can watch the status of a list of "buddy" phones. This can be used in a boss - secretary setup; however, My Status has to be updated manually on the phone with two exceptions: Do not disturb and in a call automatically update My Status. (This feature uses SIP/SIMPLE and should be compatible with Windows Messenger and MSN Messenger according to the Polycom manual. I have not been able to test this yet; however, different versions of Windows or MSN messenger use different protocols and therefore might or might not work. See also HowTo configure Windows Messenger with sipX).
Note: After release 1.5x Polycom changed the default phone configuration to disable presence and messaging functions. You must manually enable presence support for each phone, by checking the "Presence" box under Features in the phone configuration. (Take care to have the version pop-up set to 1.6 rather than 2.0.)
Note: In release 2.0x and above Polycom changed the way presence operates in a way that makes it incompatible with sipX 1.6 and earlier, so presence support does not work in sipX 3.6 with Polycom 2.0+ firmware. This problem is fixed somewhat in sipX 3.8, which incorporates explicit BLF support (See Roadmap: sipXecs 3.8 Busy Lamp Field (BLF)). Further refinements are planned for sipX 3.10 (See Roadmap: sipX 3.10 Busy Lamp Field (BLF)).
Note: The presence indication can be used in conjunction with the Xten eyeBeam softphone. With eyeBeam you can create a buddy list, which allows the state of the users to be represented. This currently only works between Polycom and eyeBeam phones.
Blind and Consultative Transfer
All the Polycom SIP phones support both blind and consultative transfer. Nothing needs to be configured server side to enable this feature.
Note: Blind Transfer is very non-intuitive using the Polycom phones for the most users. You would expect that after pressing Transfer and dialing the number you want to transfer the call to, you could hang-up and the transfer would complete automatically. This is not the case! The correct sequence is: Press Transfer, then a new softkey Blind appears in the display. Press Blind, then dial the new number and hang up. The transfer now completes succussfully as intended.
Multi-Line Appearance
Also, the Polycom phones support multi-line appearance. If a line is in a call and a second call comes in on that line, the user is given the option to put the first call on hold and pick up the new call coming in. Even though on the phone menu it offeres Conference, I was unable to actually conference the two calls together.
Conferencing
The Polycom phones support local conferencing of up to three parties (including the station doing the conferencing). No configuration server side is necessary to enable this feature as it is done by the phone. It is also possible to define an external media server or conference bridge. Go to <Phone Configuration>, SIP and scroll all the way down.
Automatic Ring Down
sipX ConfigServer allows you to configure an automatic ring down address for the Polycom phones, which is useful for e.g. a lobby phone. Picking up the handset will automatically connect you to a predefined extension, such as the operator.
Firmware Upgrade (SoundPoint IP300, IP301, IP430, IP500, IP501, IP600, IP601, IP650)
sipX Configuration Server provides a GUI based mechanism for firmware upgrades.
Note: Polycom Support only distributes firmware upgrades to certified resellers. It might be possible though to find the necessary files on the Internet. Otherwise, please contact your local reseller.
Polycom Configuration Files
ConfigServer creates a sub-directory per Polycom phone in the TFTP root of the sipX server that includes all the necessary configuration information. These files are auto-generated by ConfigServer and with the exception of adding a new firmware image these files should not be changed manually.
Optional - Firmware Images (in TFTP root): bootrom.ld BootROM image file bootrom.ver BootROM version file only needed for BootROM prior to 2.6.1 sip.ld SIP firmware image file sip.ver Version number of SIP image
Phone Configuration Files: <mac.xxxx>/ipmid.cfg Main config file (integrated with sip.cfg starting with SIP 1.5.1) <mac.xxxx>/sip.cfg SIP configuration file <mac.xxxx>/phone.cfg Phone configuration file <mac>.cfg Basic per phone config file that tells the phone where to find the other files phone1.cfg Example file only; rename before use
Polycom Phone Directory Configuration
The Polycom IP300 / IP500 / IP600 phones maintain a local and phone specific contact directory that can hold a fairly large number of records dependent on available memory (see note below). The directory can be downloaded from the TFTP server and edited locally. Contact information from previous calls may be easily added to the directory for convenient future access. The directory is the central database for several other features including speed-dial, distinctive incoming call treatment, presence, and instant messaging.
Using the phone top interface the directory is accessible by pushing the Directories key. The following configuration files are used to load directory information from the TFTP server:
000000000000-directory~.xml sample directory file provided by the firmware package (needs to be renamed before use) 000000000000-directory.xml phones with an empty directory will upload this file as an initial directory <mac>-directory.xml phone specific directory file
The user of the phone can edit the directory contents at will. Changes will be stored in the phone�s flash file system and backed up to the TFTP server copy of <mac>-directory.xml. When the phone boots, the TFTP server copy of the directory, if present, will overwrite the local copy. Since the TFTP server only allows the phone to write files that already exist on the server and have global write permissions, make sure you create the file first.
On the sipX server: cd /var/sipxdata/configserver/phone/profile/tftproot cp 000000000000-directory~.xml <mac>-directory.xml chmod 666 <mac>-directory.xml
Tip: Start with an empty directory on the phone. Create the file <mac>-directory.xml based on a sample file with one or two entries in the TFTP root directory of the sipX server. Make sure the file has write permission set for everybody (chmod 666 <mac>-directory.xml). Reboot the phone and check whether the directory on the phone has been populated during the boot sequence. Add an entry using the phone top interface. Press all four arrow keys simultaneously to force the phone to upload the modified directory file to the TFTP server. Check the file on the server (path: /var/sipxdata/configserver/phone/profile/tftproot) to make sure the operation completed successfully.
Tip: Using an XML-Editor such as Bluefish on Linux you can create a template for a company wide directory file. Individual users can then add their own contacts using the phone top UI.
Note: By default the phone is configured to store directory records in non-volatile (FLASH) memory. Dependent on what model phone you have (2 MB or 4 MB of FLASH), this limits the number of records to about 20. In the sip.cfg configuration file and using the XML tag <directory> the phone can be configured to store directory records in volatile (RAM) memory instead, which significantly increases the number of records that can be stored. This configuration feature is currently not supported by sipX but might be added to ConfigServer in a future release. You cannot change it manually in sip.cfg as ConfigServer will override this setting next time a profile is pushed to the phone.
Sample <mac>-directory.xml file:
<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<directory>
<item_list>
<item>
<ln>Doe</ln>
<fn>John</fn>
<ct>1001</ct>
<sd>1</sd>
<dc/>
<ad>0</ad>
<ar>0</ar>
<bw>0</bw>
<bb>0</bb>
</item>
</item_list>
</directory>
Speed Dial Numbers
Entries in the directory can be linked to the speed dial system. The speed dial system allows calls to be placed quickly from dedicated keys as well as from a speed dial menu. The first few speed dial entries are mapped to unused line keys.
The XML tag <sd> (see example file <mac>-directory.xml above) in a directory record allows to specify the speed dial index (a number between 0 and 40) for a directory entry. The speed dial index can also be set using the phone top user interface: Press Directories, select Contact Directory amd chose an entry to edit.
Note: To access the speed dial system press the up arrow key on the phone key pad.
Security Note
TFTP does not have any provision for authentication and therefore requires files to have public read permissions. In addition, files that the phone writes back to the server, such as the phone's directory as well as log files, require public write access. This represents a significant security concern as anybody can read and write individual user's directory (phone book) information.
FTP with one global password is slightly better but still not great. Setting up FTP servers with separate passwords is possible, but you'd have to enter each password by hand into the phone's configuration and go through the hassle of setting up the FTP accounts.
It's suprising that phones do not have tighter security. The only truly secure way is to have individual certificates on the phones and although I think some phone's are starting to support this, that's still pretty advanced.
Changing Polycom Profile Templates
If the setting you want to change is not available from the web UI. You have two options, and it to the Web UI or hardcode the values in the template. Either way, you'll have to become familiar with polycom's settings format: Polycom 1.6.x Configuration File Format
Option 1: Hardcoding Polycom Templates
This is easiest of two methods but means all phones will be configured with the exact same settings you wish to control.
In the /etc/sipxpbx/polycom/mac-address.d directory you will find 2 files, sip.cfg.vm and phone.cfg.vm. The format of these files is based on the combination for 2 different file formats. First, the default settings for Polycom phones as described Polycom Admin guide mentioned above. Second , a scripting language call velocity. Find the configuration settings you want to alter, be sure you do not create invalid velocity syntax or xml file, make your edits and regenerate profiles for all your phones.
Option 2: Modifying the Settings Descriptions and Updating Polycom Templates
This will add the settings you wish to control to the web UI so you can alter the setting for each phone or each group of phones.
- Step 1: Read and follow Option 1:Hardcoding Polycom Templates mentioned above but instead of hardcoding values, insert velocity syntax.
- Step 2: Describe the settings in the polycom settings descriptor files that match the expected settings you've created in step 1. There are 2 settings descriptor files for polycom contained in the following directory:
/etc/sipxpbx/polycom. The first setting descriptor file is for phone wide settings and calledphone.xml. The second is for line specific settings and calledline.xml. You can describe on/off settings, lists of items, numbers w/valid ranges and all sorts of settings. See SipX ConfigServer settings.xml for settings language reference.
Example
To help you become familiar with the velocity syntax and settings descriptor files, I'll walk through an example for controlling the persistance of headset volume. You may have noticed that each each time you pickup the handset or headset, the volume has changed. Polycom hardware let's you keep the volume at the same level eachtime to make or recieve a call.
NOTE: These settings are already available from web ui.
- First open polycom template file in a text editor, find the settings on volume and make the following edits:
/etc/sipxpbx/polycom/mac-address.d/sip.cfg.vm
<volume
voice.volume.persist.handset="0"
voice.volume.persist.headset="0"
voice.volume.persist.handsfree="1"
/>
To this:
<volume
#set ($group = $cfg.EndpointSettings.getSetting('voice').getSetting('volume'))
#foreach ($setting in $cfg.getSettings($group))
voice.volume.${setting.ProfileName}="$!{setting.Value}"
#end
/>
- Second, because volume is a phone-wide setting open the phone settings descriptor file in a text editor and make the following addition.
/etc/sipxpbx/polycom/phone.xml
<group name="volume">
<label>Volume Persistence</label>
<description>The user's selection of the receive volume during a call can be remembered
between calls. This can be configured per termination (handset, headset and
handsfree/chassis). In some countries regulations exist which dictate that receive
volume should be reset to nominal at the start of each call on handset and
headset.</description>
<setting name="persist.handset">
<type>
<boolean/>
</type>
<value>0</value>
<description>If checked, the receive volume will be remembered between calls. If set to
0, the receive volume will be reset to nominal at the start of each
call.</description>
</setting>
<setting name="persist.headset">
<type>
<boolean/>
</type>
<value>0</value>
<description>If checked, the receive volume will be remembered between calls. If set to
0, the receive volume will be reset to nominal at the start of each
call.</description>
</setting>
<setting name="persist.handsfree">
<type>
<boolean/>
</type>
<value>1</value>
<description>If checked, the receive volume will be remembered between calls. If set to
0, the receive volume will be reset to nominal at the start of each
call.</description>
</setting>
</group>
Restart entire system, or just sipXconfig and you should see the new settings appear in you Web UI for phones and phone groups. If you make changes, you're changes are captured in the database and when you generate new polycom phone profiles, they'll contain your new settings.
Contribute you settings to SIPfoundy
Mail your modified files to sipx-dev@sipfoundry.org and describe what settings you added. By contributing your new settings, you will not have to make your customizations each time you upgrade your system and others may enhance your initial contribution making it better.
Resetting Polycom Phones to Default
Here are some details on what happens when you invoke the various options under "Reset to Default". (Menu - 3 - 2 - <password> - 1 - 4.)
These are described in Polycom Quick Tip 18298, but some important details are missing. [1]
Changes made through the Phone's Menu options or Web GUI or are considered "Overrides", and are cached locally on the Phone. Many of these, such as Line Registration and other SIP settings, overlap with the contents of the configuration files.
Two things may come as a surprize to you:
1. Overrides have precedence over the configuration file contents. 2. Overrides are not removed when you invoke "Reset Device Settings" under "Reset to Default".
When you change a Polycom's Boot server address from one sipXecs server to another, you might expect it to pick up the new configuration files and start working seamlessly. But you can get very unexpected behaviour if the Polycom has locally cached Overrides of SIP settings.
Even if you invoke "Reset Device Settings" the Overrides will not be cleared.
To clear the Overrides you need to invoke "Reset Local Config" under "Reset to Default". You could instead invoke "Format File System" which will erase everything except the BootROM, but that is more difficult to recover from.
Recommendations:
1. Clear the Overrides before deploying any Polycom that has been used before. 2. Avoid using a Polycom's Menu options to change anything under "Advanced" with the exception of "Network Configuration" and "Change Admin Password". 3. Avoid making any changes under a Polycom's Web GUI.
