Grandstream Networks GXE502X User manual

GXE502X User Manual 11/2/2009
1
GXE502X Users Manual
For Release 1.0.1.50

GXE502X User Manual 11/2/2009
2
Table of Contents
1. Introduction..................................................................................................................................4
2. Extension Configuration.............................................................................................................4
3. Trunk Configuration ..................................................................................................................10
4. Conference Bridge.....................................................................................................................23
5. Hunt/Ring Group Configuration ...............................................................................................25
6. Auto-Attendant Configuration ..................................................................................................27
7. Call Queues Configuration........................................................................................................32
8. System Configuration................................................................................................................34
9. Advanced Options Configuration.............................................................................................50
10. Reset & Reboot ......................................................................................................................63
11. Status......................................................................................................................................64
12. Reports....................................................................................................................................66
13. Voicemail Configuration........................................................................................................74
14. Personal Web Portal ..............................................................................................................76
Appendix A: Complete Voicemail Flow Chart ..............................................................................78
Appendix B: Template File..............................................................................................................79

GXE502X User Manual 11/2/2009
3

GXE502X User Manual 11/2/2009
4
1. Introduction
Thank you for purchasing the Grandstream GXE502X IP-PBX, a SIP-based, affordable, feature-rich
converging communication platform designed to meet the communication requirements for small to
medium sized enterprises.
The GXE502X provides the cutting-edge IP-based communications that businesses demand while
leveraging existing infrastructure and providing a smooth transition into IP telephony. Based on open-
standard SIP, the GXE502X can easily integrate into and interoperate with other components of your
existing communications network while providing a rich set of features to reduce costs and increase
productivity. Built-in FXO and FXS ports enable the GXE502X to interface with analog lines and
devices while concurrently registering to SIP trunks and SIP-based trunk gateways to maximize
available communications resources. The PoE (power over Ethernet) capability (LAN port) and low
power consumption (15Watt Max.) solidify the GXE as an environment friendly solution (compared to
similar PC based systems). This also ensures non-stop communication even during power outages
(necessary UPS required)
The GXE502X also includes an Express Setup wizard and an intuitive user interface that allows users
to quickly configure extensions. .. Voicemail, fax mail, faxmail-to-email, conference bridges and other
enhanced features can be turned on with minimal effort via the GXE502X web configuration interface.
The GXE’s broad feature set, ease of operation and affordable price range make it ideal for any SMB.
Equipment Packaging
The GXE502X IP-PBX package contains:
1) One GXE502X IP-PBX unit
2) One 12 Volt power adapter
3) One Ethernet cable
2. Extension Configuration
Phone Extensions
The front end menu of the GXE web interface is the Phone Extensions menu. This menu and its
several different sub menus allow users to configure and add extensions with ease.
Extensions for internal users can be created from the Express Setup menu of a factory-default
GXE502X or from the Phone Extensions menu at any time. The extensions can be local users
(shown in Green in the Extensions Directory) in the internal network, or remote users connecting over
the open Internet (shown in Blue in the Extensions Directory). This flexibility allows GXE502X users
to bring their office extensions virtually anywhere that has a broadband Internet connection. This
gives users easy access to personnel (home office and road warriors alike) without incurring any
PSTN toll charges.

GXE502X User Manual 11/2/2009
5
- Express Setup
The Express Setup process allows users to automatically configure extensions by auto-provisioning
Grandstream phones1. This sub menu will appear ONLY on factory default GXE units or after
performing a factory reset. Once the Express setup is executed, the sub menu will disappear.
Grandstream highly recommends running the Express Setup even though the user might not be
ready to install any phones at the moment. When we run the Express Setup the GXE will auto-
configure the extensions for the conference bridges, FXS ports and more importantly it will create a
digit map under the internal call routing profile. We will explain call routing profiles in a different
section but they are a key element to allow transparent communication.
If you are not going to configure any phones at the moment you can start the auto-provisioning and
then stop it a few seconds after so that the GXE can create all these items.
-Using the Express Setup wizard to create extensions and auto-provision phones.
Please refer to the GXE502X Quick Start Guide for detailed instructions on setting up your GXW with
the Express Setup wizard.
The Express Setup Wizard will be the initial point of configuration for your telephone system. The
GXE needs to know these values first so that it can properly configure the system.
Extension Length:
All extensions will have the number of digits specified here. Please select a digit from 3 to 6.
For example if you select 3, then the extensions will be like 101,102, 103…199.
If you select 4, then the extensions would be like 1001, 1002, 1003….1999.
For advanced users: You can create different digit maps in the internal call routing profiles with
different extension lengths.
Extension Digit Prefix:
All extensions will begin with this digit.
Other digits can be used for peer systems, trunks or feature access codes.
For example if you select 6, then the extensions will be like 602,605,611.
If you select 3, then the extensions will be like 307,356,322.
For advanced users: You can also setup other local extensions with different prefixes in the in the
internal call routing profile.
1Auto provisioning is currently only available for Grandstream phones

GXE502X User Manual 11/2/2009
6
After you input these two numbers, you will find the following screen:
For this example I selected 6 to be my extension prefix and 4 to be my extension length as well.
As you can see the GXE has preconfigured some of the extension numbers that will take effect only
after we start the auto-provisioning. If you wish to change any of these numbers please do it at this
point. If you want to modify the extension Prefix, press the "Prev" button, otherwise press the "Finish"
or the "Finish and Start auto provisioning" buttons; press “Finish" if you do not want to start auto
provisioning, the GXE will reboot by itself after you press this button. Press the "Finish and Start auto
provisioning" button to begin the Auto provisioning process.
The extension numbers that you see in the example are as follows:
Extension Number for Phone/FAX Ports: These are the two FXS ports labeled TEL1 and TEL2 in
the back of the GXE. You can connect analog phones or fax machines to these ports. Additionally
if you have the GXE on PoE and a PSTN line connected they can be used as your lifeline in case
of a power outage.
Extension Number for Conference Bridge: These are the extensions that correspond to each
conference room in the GXE. The 5024 model comes with 2 conference rooms and the 5028
model comes with 4 conference rooms. So each conference room is in reality a virtual extension.
We will discuss more about the conference bridges later on.
The starting and ending extensions for express provisioning will determine which range of
extensions will be provisioned if they are connected locally, using a switch, to the LAN port of the
GXE.
The extension number for the operator is simply created automatically as the first extension since
this is usually the easiest number to remember and typically used by an operator or secretary.
Once you get the auto-provisioning running you should see the screen below.

GXE502X User Manual 11/2/2009
7
When you are done with the auto-provisioning, simply click on the “stop auto provisioning and done”
button. Then click on the OK button on the pop up screen to finish. The GXE will then auto-reboot to finish
the configuration process
- Extensions Directory
The Extensions Directory sub menu allows user to add, batch add, modify, and delete extensions,
after the Express Setup process.
-Using the Extensions Directory sub menu to create, modify, view, and delete extensions:
The Extensions Directory section displays all SIP phone extensions configured on the GXE502X.
This page also displays the IP address, firmware version and registration status of the end points.
The following actions can be performed in this section:
Add an extension: Clicking on the Add One Extension button will load the extension configuration
page. This page allows users to set all of the extension’s settings. Click on the Submit button to
add the extension or Cancel to abort the add extension process.
Configuration Field Explanations:
1) User Name: Name of User as shown on the Extensions Directory screen
2) Department Name: The Name of the Department that the extension belongs to. For example
Accounting, Sales, Shipping, Support, etc.
3) Extension: The Extension number. This is usually the SIP user ID in the IP phone or ATA.
4) Privilege: This is used to determine the storage quota in flash memory for voicemail/video
mail/fax mail, based on a percentage of the total space.
5) SIP Password: The password required to allow phones to register to GXE502X. This
password must be synchronized between the GXE502X and registered SIP end points (i.e.. IP
Phone & ATAs).
6) Voicemail Allowed: Toggles voicemail on and off for this extension.
7) Ring Attempts before Forward to Voicemail: Configures the amount of rings, in seconds,
before calls are forwarded to voicemail. (When voicemail is enabled)
8) Fax mail Allowed: Enables the fax mail feature for this extension.
9) Forward Voice/Fax mail to Email: This is the email address where the GXE will forward all
voicemails and fax mails.
10) Password to Retrieve Voicemail/Fax mail: The password required to retrieve Voicemail or Fax
mail.

GXE502X User Manual 11/2/2009
8
11) Call Forward2: Toggles Call Forwarding on and off. The factory default setting is OFF.
12) Call Forward To: The phone number the call will be forwarded to. It could be another
extension or a phone number.
13) Call Forward Rule: Determines the conditions under calls are forwarded.
14) Time for No-Answer-Forwarding: This is the timer that once expired will forward the calls to
the call forward number. Please make sure this timer is shorter than the voicemail timer if the
extension has Voicemail enabled.
Advanced Options:
15) RTP Port Detection: Toggles the GXE’s ability to detect RTP ports for remote extensions. This
field is set to “Yes” on factory default units.
16) Do Not Disturb: When enabled, calls will go directly to voicemail (when VM is enabled). Calls
will be rejected if voicemail is disabled.
17) Reboot Peer: When set to “Yes”, this feature will reboot registered (online) Grandstream
endpoints when the “Submit” button is clicked. The GXE502X can force extensions to reboot
via this method. This is the check-sync method used in Notify messages.
18) Session Keep Alive: This feature lets you configure different methods to keep the session
alive: None, Update, Re-Invite or Session Timer/Automatic. The factory default value is
Automatic. Grandstream recommends using the default setting.
19) Session Expiration: Please refer to RFC 4028 for detailed information
20) Min-SE: Please refer to RFC 4028 for detailed information
21) Call Routing Profile: These are the profiles the extension will be using to place outbound calls.
The administrator can select multiple call routing profiles from the available list to enable the
extension in particular to be able to use them. For example the administrator could have
created a local, cell phone, international and peers call routing profiles so he can decide what
kind of call each extension can make. Simply select or unselect each profile from the list to
activate them. Remember to click on the submit button.
22) Authorization Profile: The administrator can assign a predefined authority profile to this
extension so that they might require a password to perform certain calls.
23)Voice mail storage time: This is the period of time in days that the GXE-502X will keep you
voice/fax mail messages stored in flash memory. After the time has passed the GXE-502X will
delete the messages automatically to save space in memory. Default 30 days.
24)Entry voicemail option: This will select in between 3 options for accessing voicemail from the
user’s extension using the feature code for voice mail.
Extension and password: after dialing the feature, code the user will be prompted to enter both
the extension number and the password to access the extension’s voicemail.
Password only: after dialing the feature code, the user will be prompted to enter his password
only to listen to his voicemails
Direct Call: After dialing the feature code, the user will start listening his voicemails bypassing
all passwords.
Batch add extensions: The Batch Add button allows you to manually add multiple extensions.
The batch add configuration page lets you specify the range of extensions to add, as well as some
general settings to apply to them. You may also modify the settings of these extensions through
the Extensions Directory page. Click on the Submit button to add the extensions or Cancel to
abort the batch add process.
2Note: Call forward can also be activated by using feature code.

GXE502X User Manual 11/2/2009
9
Modify an extension: If you want to modify any of the extension’s properties simply click on the
extension number under the extensions directory page. To reboot the Grandstream phone/ATA
(remotely), set the Reboot Peer setting to “Yes”, then click on the Submit button to modify and/or
reboot the extension or Cancel to go back to the extensions directory page.
Delete: Check the square box to the left of the extension you wish to delete. You will be prompted
for confirmation via a dialog box; click OK to confirm or Cancel to go back to the extensions
directory page. You can select multiple boxes and delete several extensions at once. Note that
this only affects extensions on the current page; extensions on other pages will not be deleted.
You will be prompted for confirmation via a dialog box; click OK to confirm or Cancel to go back to
the extensions directory page. You can also select the All checkbox to automatically select all
extensions in the displayed page.
-Busy Lamp Field and Message Waiting Indication
Busy Lamp Field (BLF) and Message Waiting Indication (MWI) support are enabled on the
GXE502X by default. New voicemail messages will automatically trigger the MWI light on phones
that support it to notify the user of new voicemail messages.
To monitor the status of other extensions in the GXE502X, configure the BLF keys on a
(Grandstream) IP phone to the extension numbers to be monitored. The BLF key configuration
can be found on the basic settings tab of your Grandstream GXP phone’s web UI. The GXE502X
will trigger the LED of BLF keys to signal when the monitored extension is idle (Green), ringing
(Blinking Red) or busy (Solid Red).
- General Settings
The Auto Provision sub menu provides users a method to Auto Provision Grandstream endpoints
after the Express Setup process.
-Auto Provisioning Extensions after Express Setup
Click on “Auto Provision” under the Phone Extensions menu. Enter the starting and ending
extensions for the phones (Make sure to enter numbers in the established digit frame structure)
before connecting your Grandstream phones.
Click the “Start” button to start the auto provisioning process. Once the Auto-Provision page starts
refreshing you may connect your phones to your switch/hub. Click “Stop” when done.

GXE502X User Manual 11/2/2009
10
3. Trunk Configuration
PSTN trunks, SIP trunks and SIP-based PSTN trunks (extended via Grandstream FXO Gateways)
can all be configured on the GXE502X in the Trunk/Phone Lines menu.
The FXO and FXS port/line electric specification parameters and settings can be configured in this
area as well. All of these will allow users of the GXE502X to make calls anywhere through multiple
trunk types.
- Internal PSTN Trunk Line Configuration (via FXO ports)
The Internal PSTN Trunk Line Configuration page lets your configure the FXO ports on the back of
the GXE502X to make and receive calls to and from the PSTN.
Compatibility with different PSTN lines can be achieved by adjusting the related parameters on the
Internal PSTN Trunk Line Parameter page. The default settings should function in most cases.
Correct technical data must be obtained from local Telco carriers and implemented on the GXE502X
before the lines will function. This data depends on the regional PSTN line characteristics.
PSTN Device List
Starting on version 1.0.1.32 the GXE support different line parameters assigned to each FXO port.
Click on the Add button to start the process of adding FXO ports to your GXE.
Add FXO Device

GXE502X User Manual 11/2/2009
11
Trunk Name: This is the name used to identify this set of trunk(s). For example: carrier A,
carrier B, Telco C, etc.
Line: Please specify the FXO ports that you want to include in this profile either one by one or
as a range. For example: 1, 2-5, 8.
Prefix: This is the prefix that the GXE will use to create the outbound call routing profile so that
it will know which trunk to grab for an outgoing call. When the user wants to make a call he will
have to use this prefix to grab this particular set of lines.
Call Routing Profile: This section will indicate the GXE where to route the calls coming into this
PSTN trunk. You will see two boxes, one with the available list and another one with the
selected call routing profiles. It is recommended to use either the play voice menu profile which
will reproduce the auto-attendant menus according to the playing rules set, or to select the
general inbound which will route the call accordingly to what the user has selected in the
general inbound conditions and values.
Advanced settings:
Click on the Advanced link to show the advanced settings options, here you will find the PSTN
Line Parameter Settings and the PSTN Port Parameter Assignments.
PSTN Line Parameter Sets:This section lets you specify the line disconnect signaling,
impedance, as well as the caller ID standard and line dialing settings.
Set Name: This is the name that corresponds to this line or set of lines to be configured. For
example a user could have 2 sets of PSTN lines from different providers with different settings.
Enable Current Disconnect: Default is Yes.
This value should be used when the PSTN provider or local Telco Carries uses a line power
drop to indicate call completion to the subscribing end point. In this case the FXO port of
GXE502X will search for a power drop with the preconfigured time frame to disconnect call
from a VoIP extension.
Current Disconnect Threshold:
This is a preconfigured value of duration for a line power drop used by specific service
providers or Telco Carriers. For example, with a configured value of 500ms, the device will
ignore any random voltage drops on the line if the duration of such a drop is less than 500ms.
The call will NOT be considered as terminated, but will clear a call if a drop is equal or more
than 500ms. This is useful to prevent unnecessary call drops in some low quality PSTN lines.
However inappropriate parameters may cause calls to drop unexpectedly or cause the line not
to release as expected. Please consult with the local Telco carrier before changing this
parameter.
Enable Tone Disconnect: Default is YES
This feature causes a Busy Tone to be used as the FXO line disconnection signal when set to
YES. It is used by some Telcos or enterprise POTS PBX.
BT Event or Busy Tone Event: This is the number of cadence cycles to be detected in order
to disconnect the call. In other words; in the cases where the PSTN provider signals a call
termination by playing a busy tone the GXE will listen to how many times these tones are

GXE502X User Manual 11/2/2009
12
played. Typically these tones are made up of 2 or 3 frequencies and are reproduced according
to an on and off cadence. The “on” cycle means that the tone is played for certain duration of
time, and the “off” cycle means that only silence is played for certain duration of time. In order
to avoid false tone detections, the BT event can be configured so that the user can specify how
many cadence cycles need to be detected in order for a call to be disconnected.
Polarity Reversal Disconnect:
If set to “Yes”, calls will be terminated after a polarity signal reversal. The default value is No.
AC Terminal Impedance Scheme:
Users can select one of two options for the AC termination scheme: Country (rough and easy)
or the Model of related Impedance values (real and complex).
Please make note that inappropriate parameters may cause echo or popping noises on the
lines.
Please consult with your local Telco Carrier before configuring these fields.
AC Terminal Impedance:
This depends on whether Country or Model is selected, either 17 Countries are selectable or
users have to select the Impedance value used by the local PSTN carriers.
Gain(dB):
This option allows the user to increase or decrease the power level of the signal (audio) sent or
received on the FXO ports. Remember that we are using a logarithmic scale in this case and
that increasing the level by 3dB will actually double the power level sent to the PSTN. In a
similar way, decreasing the power level by 3dB will cut in half the power level of the signal
received.
Transmit (TX) to the PSTN: This controls the power level sent out to the PSTN.
Receive (RX) from the PSTN: This controls the power level received from the PSTN.
Jitter Buffer Type:
This option allows the user to toggle between an adaptive or fixed jitter buffer.
This will help in eliminating jitter in the communications by buffering some packets before
transmitting them.
Call Progress Tones: This feature lets you configure tones to match the FXO line regional
settings.
In certain countries, the carriers or central office will send special tones to indicate certain events
on the PSTN side. For example, when a call is disconnected from the remote side the other side
will hear a busy tone indicating that the call is terminated. Also, a dial-tone is used by the central
office to signal that it is ready to receive digits and process the call. Users can pre-configure these
tones in the GXE502X. However users must know the correct frequency values and cadences of
these tones from the PSTN carrier. Wrong parameters may cause the GXE502X to malfunction
due to incorrect tone settings.
In other words, the user MUST configure the call progress tones in order for the GXE to work
properly. For more information about troubleshooting this please refer to the HELP section under
this menu on the GXE’s web UI.
Here is an example for the syntax for a busy tone in the U.S.A:
Syntax: f1=freq@vol, f2=freq@vol, c=on1/off1-on2/off2-on3/off3; [...]
(Note: freq: 0 - 4000Hz; vol: -30 - 0dBm; c: (cadence) in msec, several cadences allowed)

GXE502X User Manual 11/2/2009
13
Default US Busy Tone: f1=480@-24,f2=620@-24,c=500/500)
Line Dialing Settings: This feature adjusts line dialing properties to adhere to the PSTN line
requirements.
DTMF Digit Length: The DTMF Digit Length is the play time of DTMF in milliseconds for each
digit dialing out from the VOIP side to PSTN
DTMF Digit Volume: The digit output power in dBm.
DTMF Dial Pause: Dial pause is the time between 2 digits when dialing out through the PSTN.
Wait for Dial-Tone: If set to Yes, the device will first obtain a PSTN line and wait for a dial
tone from the central office. After obtaining the dial tone, the digits dialed will be sent to the
central office.
Minimum Delay Before Dial: This parameter lets users configure a variable delay before
dialing the first digit into the PSTN.
Caller ID Standard: This feature configures caller ID handling to match local PSTN settings.
Caller ID scheme: This feature lets you select the model used by local PSTN provider. Please
check with local PSTN carrier for configuration information. Current supporting including:
Bellcore, ETSI_RING, ETSI_TAS, DTMF or NTT
FSK Caller ID Minimum RX Level: This is an adjustable threshold value for the Caller ID
signal strength received to help the device to recognize Caller ID from different networks. (-96 -
0dB. Default -40dB)
FSK Caller ID Seizure Bits: This field lets you choose a value of alternating bits (zeros and
ones) used for CID channel signaling. The Default is 200.
FSK Caller ID Mark Bits: This feature lets you choose a value for the frequency modulated
signal of alternating bits (zeros and ones) used for synchronization. Default 150.

GXE502X User Manual 11/2/2009
14
Automatic Detection:
This section will be of great help to the user in order to determine the impedance, call progress
tones and current disconnect settings on the FXO ports of the GXE.
Remember that the impedance on the FXO ports must be as close as possible to the impedance
on your PSTN lines in order to reduce echo. If we do not match the impedance then we will have
problems with echo on calls, calls getting dropped and faxes not getting received.
Call progress tones will tell the GXE when the line has dial tone, when the line is getting ringback
tone and when to disconnect the line based on the busy tone heard at the end of a conversation.
These tones vary from country to country and the user must configure them according to the
geographical location where the GXE will be installed.
Current disconnect is a method used in a few countries, including the USA, that removed the
current from the line for a split second following the end of a conversation. This method is used to
determine when the line has been disconnected.
If the user ignores these settings the GXE may not work properly. No country in the world should
be using current disconnect and tone disconnect simultaneously, but these settings depend
completely on the central office. Impedance on the PSTN lines could vary from one to another as
the physical medium is susceptible to many changes.
Before starting the auto detection please make sure that the wires in your PSTN line are in the
correct order and that they work properly connected to an analog phone directly. You will need to
know the PSTN numbers corresponding to each wire.
Caller FXO (detected FXO): This will be the FXO port that we will detect the values for. Please
insert the line number (FXO port in the back of the GXE) and the PSTN line number for it. In other
words if we have a PSTN line connected to FXO port 1 with the phone number 469-241-0100. We
should type in 1 for the line number and 4692410100 for the PSTN number.
The important thing is to enter the PSTN numbers as if they were to dial each other locally
(because they will). So you do not need to include country codes or area codes if your PSTN
provider does not require you to input them. Do not include any dashes.
Called FXO: This is the FXO port that the caller FXO will call in order to run the tests. Simply input
the FXO port number and the PSTN number as explained above.
You can only test for one parameter at a time!
This means that you can only test for impedance or current disconnect or tone disconnect at a
single time.
Once you start the test all the services on the GXE will be stopped, and the user will have to
reboot the GXE for the services to restart.
Select only one of the tests and then click on start detect. The auto detection process will take a
few minutes; once it is done it will pop up a window with the test results. You must make a note of
these results and then apply them yourself manually after reboot. In other words the results do not
take place automatically!

GXE502X User Manual 11/2/2009
15
- Internal Phone/Fax Port:
In the Internal Phone/Fax Port section, the electrical and DTMF signaling settings of the two FXS
ports on the back of the GXE502X can be configured for compatibility with local analog phone/fax
devices. The extension numbers for these ports are usually set in the Express Setup page.
DTMF Transport Type: This field is used to specify the method of sending DTMF digits to the
VoIP side. There are three possible options: in-band (audio), via RTP (RFC2833), or via SIP
INFO. It is possible to select several methods. For example:
ch1:T=<2833/audio/signal>;ch2:T=<2833/audio/signal> . Default: ch1-2:T=2833.
SLIC Setting: subscriber line interfacing circuit configuration. This field should be configured
based on local Telco standard and analogue phone/fax type.
Caller ID Scheme: The following schemes are currently supported: Bellcore/Telcordia, ETSI-
FSK, ETSI-DTMF, SIN 227 –BT, & NTT Japan.
Polarity Reversal: If set to “Yes”, the polarity will be reversed upon call establishment and
termination; this is usually used for billing purposes. The Default is No.
Current Disconnect: If set to “Yes”, this feature will use a momentarily voltage drop to signal
remote party disconnects.
Current Disconnect Duration: A configurable time threshold where the FXS port will drop off
voltage on the line to indicate to the local party that the call is disconnected from the remote
side. (100-10000 ms. Default 100 ms)
Hook Flash Timing: The time period when the cradle is pressed (Hook Flash) to simulate a
FLASH. Adjusting this time value correctly can prevent unwanted activation of the Flash/Hold
and automatic ring-back. These values are important when adjusting the Hook Flash timing for
FXS gateways registered to the GXE
Call Routing Profile: The user can select here which call routing profiles will be allowed on
these extensions for outgoing call purposes. By default the Internal Call and General Inbound
are added automatically, the user can later restrict them to only Internal Calls for example by
removing the General Outbound profile.
Voice mail storage time: This is the period of time in days that the GXE-502X will keep you
voice/fax mail messages stored in flash memory. After the time has passed the GXE-502X will
delete the messages automatically to save space in memory. Default 30 days.
Entry voicemail option: This will select in between 3 options for accessing voicemail from the
user’s extension using the feature code for voice mail.
Extension and password: after dialing the feature, code the user will be prompted to enter both
the extension number and the password to access the extension’s voicemail.
Password only: after dialing the feature code, the user will be prompted to enter his password
only to listen to his voicemails
Direct Call: After dialing the feature code, the user will start listening his voicemails bypassing
all passwords.

GXE502X User Manual 11/2/2009
16
When finished, click on the Submit button to save your changes.
- SIP trunk
SIP trunks can be viewed, created, or modified on the SIP Trunk Configuration page. All configured
SIP trunks as well as their details and current status are displayed. Please see the following detailed
explanations of the fields on the SIP Trunk Configuration page:

GXE502X User Manual 11/2/2009
17
Add: Clicking on the Add button will display the Add SIP Trunk page. This page allows users to
enter the SIP trunk account and registration information.

GXE502X User Manual 11/2/2009
18
oTrunk Name: Enter a descriptive name in this field to identify the configured trunk.
oTrunk Active: Set to enable to activate the SIP trunk to be configured.
oSIP Server URL: The SIP server IP or URL provided by your SIP trunk service provider.
oOutbound Proxy URL: The Outbound Proxy or Session Border Controller IP or URL
provided by the SIP trunk service provider.
oAccount Name: Lets you specify a name to identify the SIP trunk service provider.
oAccount ID:The Account ID provided by your service provider, usually the VoIP line/trunk
number.
oAuthenticate ID: The Authenticate ID used by your VoIP or SIP trunk service provider.
oPassword: The password of the account provided by your service provider.
oMax. Concurrent Calls Allowed: The maximum number of concurrent calls allowed by your
SIP trunk service provider. This depends on whether dynamic or static SIP trunks are used.
The number can range from 1 to 20 given that there is enough available bandwidth on the
physical link pipe.
oDial Prefix: Sets the Dial Prefix digit(s) required to make an outbound call through this
trunk. When the call is sent out via this trunk, the dial prefix digits will be removed. By
default these digits will automatically create a new digit map on the General Outbound
profile. The new digit map will include dial prefix followed by a dot (.); these dial prefix will
be removed by the digit manipulation field and routed according to the trunk value. For
more information please refer to the call routing section.

GXE502X User Manual 11/2/2009
19
oRegistration Retry Interval: The expiration time interval for the GXE502X to retry or update
registration. This is usually regulated by the SIP server configured here. Reducing this time
interval will keep registration more synchronized with the SIP server but with the cost of
more registration packets sent to server. The default value is 600 seconds or 10 minutes.
oSIP Transport: This will be the transport method used by the SIP Trunk. This method is
given by the SIP trunk provider. The options are UDP (default), TCP or TLS.
oHeart Beat: if set to Yes, the GXE502X will send a SIP OPTIONS Request frequently to
check whether the SIP Proxy is still online. If the Proxy does not support by answering with
a SIP 200 OK then the GXE502X will consider this trunk as disconnected. This feature is
also used to keep a UDP port open when the GXE502X is behind a NAT firewall.
oSession Keep Alive: This field allows users to enable or disable usage of the session timer.
This is enabled by default to: Automatic/Session Timer.
oSession Expiration: The session timer enables SIP sessions to be periodically “refreshed”
via a SIP request. If enabled, before the session interval expires, the GXE502X will send a
SIP re-INVITE message to the SIP server, if this message is not replied the session will be
terminated. This will avoid stalled sessions occupying the channel/bandwidth. Session
Expiration is the timer (in seconds) at which the session is considered timed out if no
successful session refresh transaction occurs beforehand. The default value is 180
seconds. Incorrect configurations could cause calls to get dropped.
oMin-SE: The minimum session expiration (in seconds). The default value is 90 seconds.
oUse DNS SRV: Default is No. Select “Yes” if the domain name of the SIP server or
Outbound Proxy is compliant with RFC2782. The SIP trunk will not function properly if this
field is not configured correctly.
oUnregister on Reboot: Default is No. If set to “Yes”, The GXE502X will first send a
registration request to remove all previous or multiple bindings by adding “*” in the SIP
Contact Header. Please use this feature ONLY if the proxy supports removal binding
requests. Otherwise the SIP trunk will not function properly.
oRegister Active: This parameter controls whether the GXE502X needs to send REGISTER
messages to the Proxy Server before making or receiving calls. The default setting is “Yes”
(for most dynamic SIP trunks).
Note:
For security reasons: If this is set to “No” (for most static SIP trunks) any incoming request
MUST match the SIP server IP or the FQDN resolution must match the sending IP address.
oCBCOM Encryption: If set to Yes, RTP will be encrypted as per the algorithms of CBCOM.
The Default setting is No.
oIncoming Calls Routed by: If set to Requested-URI, then all calls are routed base on the
Request URI field in SIP Invite. If set to To-URI, then calls will be routed base on To-URI,
this is normally the case when To-URI matches the DID assigned to this SIP Trunk.
oReInvite Delay: This option allows the user to delay the SIP ReInvite sent out by the GXE in
order to finish establishing the call. The delay can be set from 0 to 4 seconds. This is very
helpful in establishing interoperability with certain service providers or platforms who cannot
handle ReInvites too quick.
oAccount ID as From Name: If set Yes, then all outgoing calls from this SIP Trunk will use
the Account ID (In this case the account name for SIP Registration) in From Header of the
SIP Invite.

GXE502X User Manual 11/2/2009
20
oEnable Video: This option allows the user to enable or disable video offering on the SDP
body of the GXE’s SIP replies. If your SIP trunk does not support video, please disable this
option to avoid any interoperability problems.
oAdd PAI Header: This option enables the insertion of the P-Asserted-Identity header in the
SIP invite that is used on outgoing calls.
oCall Routing Profile: This section will indicate the GXE where to route the calls coming into
this SIP trunk. You will see two boxes, one with the available list and another one with the
selected call routing profiles. It is recommended to use either the play voice menu profile
which will reproduce the auto-attendant menus according to the playing rules set; or to
select the general inbound which will route the call accordingly to what the user has
selected in the general inbound conditions and values.
oDID Switch: This field is used to route incoming calls when a DID (Direct Inward Dialing #)
is used instead of the account ID. It is very helpful when there are several DIDs related to
the same (static) trunk provided by your SIP Trunk Service Provider.
Note:
All SIP requests are verified by the GXE502X so the User Part of the SIP URI has to match
the account ID or the DIDs. For outgoing calls when DID switch is enabled, the DID number
in the list will be used in the From, Contact and PAI (P-Asserted-Identity) value.
If a prepend prefix is configured, it will also be used as prefix for the DIDs.
For example, if the DID is [phone] and for outgoing calls the service provider
requires “+”, then configure the prepend prefix to “+”and add the DID 16175669300. Then
[phone] will be used to match both incoming calls and also outgoing calls.
Click the Submit button to add the trunk.
Modify: Click the SIP trunk name to modify the trunk’s parameters. The SIP trunk details page
will be displayed, allowing users to modify all of the SIP trunk’s settings. When done, click the
Submit button to save all the changes.
Delete: Click the Delete button on the bottom of the row to delete the SIP trunk. Users will be
prompted for confirmation via a dialog box; click OK to confirm or Cancel to go back.
One Button Provisioning
“One-Button-Provisioning” for SIP Trunk provisioning allows the user to provision Grandstream
Certified SIP Trunk Providers with just a few steps:
Fill in the contact information
Select the certified SIP Trunk service provider
Click on “submit” button to activate the service
Note: