Polycom SoundPoint IP30x

Polycom SoundPoint IP30x
Polycom SoundPoint IP30x, IP430, IP50x, IP60x, IP4000, IP650
Important Notes
WARNING: The IP-601, IP-650, and IP430 power supplies are NOT
compatible with any of the previous models. Prior to the these models,
power supplies could be interchanged between units. If a power supply
from any other unit is used on the models listed above, it will reboot unpredictably. However, if a power supply from the 601,430, or 650 is used
on any other model, permanent damage could result to the unit.
Check the SIP 3rd Party Validation Website for current validation status.
The SIP 3rd Party Component Validation Website can be viewed at:
http://testlab.inin.com or http://testlab.vonexus.com
BootROMs 3.x and newer are signed. Once the 3.x BootROM is installed,
the phone cannot be downgraded to the 2.x series BootROM.
As of firmware v1.5.x, there is no longer an ‘ipmid.cfg’ this has been
merged with the ‘sip.cfg’
As with any Polycom firmware upgrade, it is not recommended to use
pre-existing configuration files. From version to version the XML tags in
the config files change. If older config files are used with newer firmware
versions the results will be unpredictable. I3 Support will insist that
appropriate config files be used with any given version of Polycom
FTP is the recommended/tested configuration option.
Due to the limited configuration options, configuration via the web
interface is not recommended or supported.
SFTP is not supported by Polycom as of the current firmware release, even
though it is a selectable menu item.
Do not mix configuration options (FTP, web interface, or the phone
menus). It is very easy for one interface to over ride another and limit
the ability to fully configure the phone until a complete reset of the phone
and configuration files is performed.
Vendor Documentation
Updated documentation can be found on Polycom’s website:
Validated Firmware Versions
IP 650 Firmware:
IP 601 Firmware:
IP 501 Firmware:
IP 430 Firmware:
IP 301 Firmware:
IP 4000 Firmware:
These phones were tested with BootROM version: 3.22.0019
All other Polycom phones models:
BootROM: 3.2.1
Not all combinations of BootROM and firmware versions are validated. The
current testing period was performed with the BootROM. It is
recommended to use firmware with this tested Bootrom.
A functional file distribution mechanism will be needed to distribute
firmware and configuration files to your phones. You can choose from
options like FTP, TFTP, and HTTP/HTTPS. Once a functional FTP server has
been setup, credentials to log into the FTP server will be required. The
phone will use this account to connect to the FTP server and download
BootROM, Firmware, and configuration files.
TFTP is the recommended server for small to midsize implementations. In
order for the Polycom phones to function as designed, the ability to write
configuration and log files back to the boot server is required. Since TFTP
provides no facility for authentication, this opens up the possibility that a
malicious user could manipulate configuration files on a standard TFTP
server running in a read/write mode. For this reason, version 2.4 of the
CIC and EIC products include a custom TFTP service on the server that has
more advanced options to enhance the security of TFTP. The ININ/Vonexus
TFTP server adds facilities like file extension filters and IP filters to create a
secure, yet simple way to support your IP phone base.
A new option available with the 1.5.x firmware is the ability to provision
the phones via HTTP and or HTTPS. Currently the ability to PUT a file back
to the web server does not exist in any off-the-shelf Windows based web
server package. In order to capture logs during the troubleshooting
process, the phones need the ability to PUT files back on the web server.
There are after market mods to allow this functionality, but these were not
tested. The preferred provisioning method continues to be FTP.
There is also an option for SFTP in the boot menu of the phone. The
protocol is not enabled yet and the menu item was overlooked with the
current firmware release.
Line Appearances: The line appearance is equivalent to a station in the
Interaction Administrator.
SoundPoint IP4000 – 1 line appearance
SoundPoint IP30x – 2 line appearances
SoundPoint IP430 – 2 line appearences
SoundPoint IP50x – 3 line appearances
SoundPoint IP600 – 6 line appearances
SoundPoint IP601 – 6/20/34/48 line appearances (6 standard, 14 on each
side-car with a maximum of three side-cars).
SoundPoint IP650 – 6/20/34/48 line appearances (6 standard, 14 on each
side-car with a maximum of three side-cars).
Call Appearances: All line appearances share up to 4 call appearances.
Using persistent connections allows the phone to handle more call
appearances than the phone is physically capable. This is done by using
the persistent connections and the Interaction Center Client. To manage
more calls than the phone is capable (for instance an operator want to
handle up to 20 simultaneous calls), check the Persistent checkbox in the
Station configuration in Interaction Administrator. The Interaction Client
can be used to manage a large number of calls while the phone will be the
audio device for the calls. The phone will show one call (from the
Interaction Center) while the Interaction Client will be used to manipulate
the calls.
A new feature introduced with the 1.5.x firmware is the ability to have a
single line appearance span multiple buttons on the phone. This allows for
a more traditional call behavior, similar to that found in many PBX’s. Calls
can be picked up, selected via the line appearance buttons. And if a call is
active on one line, a second call will “roll-down” to the next configured line
appearance. Details on how to configure this option are detailed in the
Polycom Administrator Guide. In Interaction Administrator set the number
of call appearances on the station to match the number of line
appearances a particular station will span.
Configuration files
The Polycom SoundPoint SIP software package consists of a binary image
file (sip.ld), a binary BootROM file (bootrom.ld), and three XML based
config files (macaddress.cfg, phonex.cfg and sip.cfg).
The XML-formatted configuration files consist of master configuration
(macaddress.cfg) files and phone specific (phonex.cfg) files. Master
configuration files identify the executable image as well as an ordered list
of phone specific configuration files to be applied to the phone.
In the event that a phone’s master config file cannot be found, it will
attempt to download a generic master config file ‘000000000000.cfg’ if it
exists. The generic master config file contains references to ‘phone1.cfg’
so this will also be applied to the phone. Here is an example of the default
master config file:
<?xml version="1.0" standalone="yes"?>
<!-- Default Master SIP Configuration File-->
<!-- Edit and rename this file to <Ethernet-address>.cfg for each phone.-->
<!-- $Revision: 1.13 $ $Date: 2004/11/26 23:30:44 $ -->
<APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="phone1.cfg, sip.cfg"
Master configuration files contain three XML attributes:
the path name of the application executable
a comma-separated list of configuration files
a comma-separated list of other required file
The order of the configuration files listed in CONFIG_FILES is significant.
The files are processed in the order listed (from left to right). The same
parameters may be included in more than one file. The parameter found
first in the list of files will be the one that is effective.
Configuration steps for use with Interaction Center
The following steps lay out the minimum steps required to use the Polycom
phones as managed SIP stations with the Interaction Center.
1. Set up the FTP server and copy all the template XML files and
application files that came with the Polycom phone into the root
directory of the FTP server. These files include,
sip.ld, 000000000000.cfg, phone1.cfg, sip.cfg, bootrom.ld, bootrom.ver
2. Create per-phone configuration files.
Create per-phone phoneXXXX.cfg and <Ethernet address>.cfg files
by using the phone1.cfg and 000000000000.cfg files as templates.
Edit the contents phoneXXXX.cfg as follows,
reg.1.displayname=”Name displayed in the SIP Messaging”
reg.1.address=”User portion of the station’s SIP address”
reg.1.label=”The button label on the phone”
reg.1.server.1.address=”IP address of Interaction Center”
Edit the CONFIG_FILES in the <Ethernet address>.cfg file so that it
references the appropriate phoneXXXX.cfg file, that is, replace the
reference to phone1.cfg with phoneXXXX.cfg.
3. Configure the phones to use the FTP server as its boot server.
a. Reboot the phone and press the Setup button that appears
during the boot screen countdown. The default password to
enter the Setup menu is “456”.
b. Use the arrow key to navigate to the Server Menu… option.
Press the Select button to enter the Server setup.
c. Use the arrow keys and Edit options to enter the IP address of
the FTP server in the Server Address: field and enter the
username and password in the FTP User: and FTP Password:
fields to match those setup on the FTP server.
Voicemail Retrieval
There are two ways to configure a Polycom IP phone for voicemail retrieval,
1. Program the Message Center (Messages button) Programming the
Message Center Connect option is a 3 step process for retrieving voicemails
but it’s initiated by pressing the Messages button on the phone. To retrieve
voicemail, the user presses the Messages button, then selects the Message
Center… option followed by the Connect soft-key.
Programming the Message Center is accomplished by setting the following
values in the phone specific configuration file.
Note: 7777 is for this example only and the actual entry should contain the
value of the IP Message Button parameter.
To access this setting, the user must,
a. Press the Messages button on the phone.
b. Select 1. Message Center …
c. Select the Connect soft-key
2. Program a speed dial button. Programming a speed dial button allows
the user to dial the voicemail pilot number by pressing a single button, but
it also requires an unused line appearance.
Programming the speed dial button is a simple process. The Polycom
phones will look for an XML file called MACAddress-directory.xml on the ftp
server when it boots. This file contains a list of phone directory entries and
speed dial entries. The Polycom distribution comes with a template file
called 000000000000-directory~.xml. Add the following entry to create a
Messages speed dial button on the Polycom phones.
<?xml version=”1.0” standalone=”yes” ?>
<fn> Messages</fn>
<ct> 7777</ct>
Note: 7777 is for this example only and the actual entry should contain the
value of the IP Message Button parameter.
The Polycom SoundPoint line supports auto-answer based on call type. For
instance, if a call is placed from the Interaction Client, the connection call to the
phone will be auto-answered. The same holds true for an ACD assigned interaction
(assuming the ACD Options are setup properly) or a connection call as a result of
opening a voicemail with the Voxform. The connection call will also be autoanswered if the pickup button is clicked in the Interaction Client, or if the pickup
option is clicked in the “toast” popup with the .NET Interaction Client.
In order to enable this functionality, find the following line in the ‘sip.cfg’
<alertInfo voIpProt.SIP.alertInfo.1.value="" voIpProt.SIP.alertInfo.1.class=""/>
And replace it with this:
<alertInfo voIpProt.SIP.alertInfo.1.value="&lt;http://localhost/AutoAnswer&gt;"
Next open up the station configuration for the Polycom phone and on the
General/Configuration page, enter “Polycom” (not case sensitive) in the
Manufacturer field. Make sure and save changes to the station.
After the above steps have been completed, reboot the phone to reload the
modified config files. Exit and restart the Interaction Client attached to that phone.
Verify operation of auto-answer in the desired scenarios.
Zone Paging (xIC 2.4 and beyond)
The Zone paging option requires that the Auto-Answer functionality
described in section be enabled. Once auto-answer has been enabled, all
phones in a particular zone will be put into a station group in Interaction
From any phone on the system dial “*901” and press a line key. Note that
you cannot press a line key then dial the “*901” as the Polycom dial plan will
interfere with the dial string.
Once you dial the “*901” you’ll be ask for the extension of the party you
wish to page. You can enter a single station extension here or the extension of the
station group.
When the call is places all phones will be paged and placed off-hook to all
the caller to announce the “page.”
Shared Line Appearances (xIC 2.4 and beyond)
This functionality will allow two physical phones to share the same line
appearance (station in IA). Along with the ability to share the line appearance the
phones will also indicate the status of line in question.
As a call is delivered to the shared line, both stations (A&B) will ring to
indicate an incoming call. As one phone in the pair answers the call (A), the LED on
the line of the idle phone (B) will glow red (IP50x phones will show horizontally
moving phone icon on the LCD display), showing the line is in use. If the user of
the idle phone (B) presses the line key, dial tone will delivered and it will be
possible to place a call from the line, even though it indicates busy.
If the call is placed on hold (A) the LED will flash red on both phones (A&B)
(IP50x phones will show phone icon with the handset turned upside down on the
LCD display). The call can then be picked up from the idle phone (B). Once this
operation is performed, the call will be connected to that phone and the LED on the
first station (A) will glow red indicating a busy status (IP50x phones will show
horizontally moving phone icon on the LCD display).
Configuration of the phones is similar to the standard setup with the
exception of a couple of changes made to the reg.x.server.x line in the phone
specific config file. Here’s and example from a primary phone:
reg.3.displayName="ip600-1" reg.3.address="8401" reg.3.label="8401"
reg.3.type="shared" reg.3.thirdPartyName="8401" reg.3.auth.userId=""
reg.3.auth.password="" reg.3.server.1.address=""
reg.3.server.1.port="" reg.3.server.1.transport="DNSnaptr"
reg.3.server.2.transport="DNSnaptr" reg.3.server.1.expires=""
reg.3.server.1.register="" reg.3.server.1.retryTimeOut=""
reg.3.server.1.retryMaxCount="" reg.3.server.1.expires.lineSeize="" reg.3.acdlogin-logout="0" reg.3.acd-agent-available="0" reg.3.ringType="2"
reg.3.lineKeys="" reg.3.callsPerLineKey=""
Notice the type is set to shared.
Here’s the line config from the other phone that is sharing this particular line
reg.3.displayName="ip601-1" reg.3.address="84011" reg.3.label="8401"
reg.3.type="shared" reg.3.thirdPartyName="8401" reg.3.auth.userId=""
reg.3.auth.password="" reg.3.server.1.address="" reg.3.server.1.port=""
reg.3.server.1.transport="DNSnaptr" reg.3.server.3.transport="DNSnaptr"
reg.3.server.1.expires="" reg.3.server.1.register=""
reg.3.server.1.retryTimeOut="" reg.3.server.1.retryMaxCount=""
reg.3.server.1.expires.lineSeize="" reg.3.acd-login-logout="0" reg.3.acd-agentavailable="0" reg.3.ringType="3" reg.3.lineKeys="" reg.3.callsPerLineKey=""
This is all the configuration required on the phone end. During testing a
numbering scheme of xxxx1 was used to note the shared lines. So if the primary
extension was 1234, then the first shared appearance would be 12341, and the
second would be 12342, etc. This is not a requirement, but something similar
should be considered to reduce confusion when troubleshooting.
Configuration in Interaction Administrator for shared lines is covered in the
2.4 documentation.
Line Roll-Down (Polycom firmware 1.5.x and beyond)
Post 1.5.x firmware, the Polycom phones have the ability for inbound calls to
“ring down,” the line keys. This behavior is very similar to legacy PBX operations
where each user had a roll-over line. If the primary line is busy, a second incoming
call would “roll-down,” to the next line key. This behavior can be configured for as
many line keys as the Polycom phone has.
Configuration for this feature is on a per line basis and can be found at the
end of each line definition in the phone specific config file. Two settings are
1). reg.1.lineKeys="3"
This tells phone phone how many line appearances each line
definition should take. In this case, once the phone has been
booted up, this line will occupy the first three line keys on the
2). reg.1.callsPerLineKey="1"
This tells the phone to allow one call per line. This can be
increased but will not mimic legacy PBX behavior. As the next
call comes in, it will ring the second line key, and then the
third, up until the maximum is reached as specified in the
above parameter.
The only requirement for this functionality in Interaction Administrator is to
configure the maximum call appearances on the specific station to match the
maximum number of line keys that you’d defined.
As calls are delivered you will be able to move between them using the line
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