Connect your PC to this port. It will then be assigned an IP address from your Router/DHCP Server. The GXW410x acts as a switch only.

Connect to the internal LAN network or Public Internet.

Factory Reset button.

Press for 7 seconds to reset the factory default settings.

Power adapter connection

Off/On switch

FXO ports to be connected to physical PSTN lines from a traditional PSTN PBX or PSTN Central Office.

Power LED

Indicates Power. Remains ON when Power is connected and unit is turned ON.

Ready LED

Remains ON after boot-up.

WAN LED

Indicates WAN port activity in the back side

LAN LED

Indicates LAN port activity in the back side

LEDs 1 – 8

Indicate status of the respective FXO Ports on the back panel

Busy – ON

Available – OFF

Profile 1

SIP Server – Set it to IP Address of GXW410x

SIP Registration – No

Outgoing Call without Registration – Yes

NAT traversal – No

Advanced Settings

STUN Server – Blank

Use Random Port – No

Advanced Settings

STUN Server – Blank

FXO lines

Wait for Dial Tone – Y or N (whichever works for your PSTN Service Provider)

Stage Method – 1

Unconditional Call Forward to VOIP:

ch1-8:444; @ch1-8:p1; ch1-8:5060++;

Channels

1-8 5060 Profile 1

Local SIP Listen port (For VOIP to PSTN calls) – 5060++

Profile 1

SIP Server – Set it to IP Address of GXW42xx

SIP Registration – No

NAT traversal – No

LAN interface

2xRJ45 10/100Mbps

LED

8 LEDs (GREEN)

Universal Switching

Power Adaptor

Input: 100-240V AC, 50/60Hz, 0.5A Max

Output: 12V DC, 1.25A

UL certified

Dimension

225mm (L) x 172mm (W) x 42mm (H)

Weight

0.29 lbs (3.5 oz)

Temperature

32~104°F

0~40°C

Humidity

10% – 90% (non-condensing)

Compliance

FCC, CE

GXW410x FX0 Analog Gateway Series

IP settings

GXW4104: 4 ports; 4 SIP accounts w/ choice of 3 SIP Server profiles

GXW4108: 8 ports; 8 SIP accounts w/ choice of 3 SIP Server profiles

Round-robin port scheduling to ensure available lines to access PSTN networks

Telephone Interface

FXO, RJ11

Network Interface

Two (2) 10/100 Mbps, RJ45

LED Indicators

Power, Video, and Line LEDs

On/Off Switch

Yes

Voice over Packet Capabilities

G.168 compliant Echo Cancellation, Dynamic Jitter Buffer,

Modem detection & auto-switch to G.711

Voice Compression

G.711U, G711A, G.723, G.729A/B, GSM

Video Surveillance

Real-time H.264 base CIF resolution

DHCP Server/Client

Switch Mode and PPPoE

Fax over IP

T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Pass-through

QoS

Diffserv, TOS, 802.1 P/Q VLAN tagging

IP Transport

RTP/RTCP and RTSP

PSTN Signaling

FXO Loop start, Current Disconnect.

DTMF Method

Flexible DTMF transmission method,

User interface of In-audio, RFC2833, and SIP Info

IP Signaling

SIP (RFC 3261)

Provisioning

TFTP and HTTP

Media

SRTP

Control

TLS and SIPS (pending)

Management

Syslog support,

HTTPS and telnet (pending), remote management using Web browser

Caller ID

Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID

Polarity Reversal / Wink

Yes (Detection only). The PSTN lines will need to be subscribed to PR service from the Service Provider.

EMC

GXW410x: EN55022 Class B, CFR Part 15 Class B, EN55024;

GXW4104: FCC, CE (in addition)

Safety

GXW410x: EN60950-1 GXW4108: UL60950-1 (in addition)

Maintenance

Web/Telnet access

Web Access

Select HTTP or secure HTTPS protocol for Web Access

Web Port

This option defines Web Port desired to use. This field is optional. Default is 80 for HTTP and 443 for HTTPS.

End-User Password

This contains the password to access the End-user Web Configuration Menu (Status and Basic Settings). This field is case sensitive with a maximum length of 25 characters.

Admin Password

Contains the password to access administrative settings other than Basic Settings and Status Page.

Upgrade/Provisioning

Firmware Upgrade & Provisioning

This radio button will enable GXW410x to download firmware or configuration file through either TFTP or HTTP.

Via TFTP Server

If selected, the GXW410x will attempt to retrieve new configuration file or new code image from the specified TFTP server at boot time. It will make up to 5 attempts before timeout and then it will start the boot process using the existing code image in the Flash memory. If a TFTP server is configured and a new code image is retrieved, the new downloaded image will be verified and then saved into the Flash memory.

Note: Please do NOT interrupt the TFTP upgrade process (especially the power supply) as this will damage the device. Depending on the network environment this process can take up to 25 or 30 minutes.

Via HTTP Server

The URL for the HTTP server used for firmware upgrade and configuration via HTTP.

For example, ttp://provisioning.mycompany.com:6688/Grandstream/1.0.0.54

Here “:6688” is the specific TCP port that the HTTP server is listening at, it can be omitted if using default port 80.

Note: If Auto Upgrade is set to No, GXW410x will only do HTTP download once at boot up.

Firmware Server Path

IP address or domain name of firmware server.

Config Server Path

IP address or domain name of configuration server.

Firmware File Prefix

Default is blank. If configured, GXW410x will request firmware file with the prefix. This setting is useful for ITSPs. End user should keep it blank.

Firmware File Postfix

Default is blank. End user should keep it blank.

Config File Prefix

Default is blank. End user should keep it blank.

Config File Postfix

Default is blank. End user should keep it blank.

Allow DHCP Option 66 to override server

Default value is No. If set to Yes, configuration file will originate from the DHCP server.

Automatic Upgrade

Choose Yes to enable automatic upgrade and provisioning. In “Check for new firmware every” field, enter the number of minutes to enable GXW410x to check the server for firmware upgrade or configuration. When set to No, GXW410x will only do upgrade once at boot up. Other options are:

“ Always check for New Firmware.”

“ Check New Firmware only when F/W pre/suffix changes”

“ Always skip the Firmware check”

Syslog Setup

Syslog Server

The IP address or URL of System log server. This feature is especially useful for ITSP (Internet Telephone Service Provider)

Syslog Level

Select the GXW to report the log level. Default is NONE. The level is one of DEBUG, INFO, WARNING or ERROR. Syslog messages are sent based on the following events:

1. product model/version on boot up (INFO level)
2. NAT related info (INFO level)
3. sent or received SIP message (DEBUG level)
4. SIP message summary (INFO level)
5. inbound and outbound calls (INFO level)
6. registration status change (INFO level)
7. negotiated codec (INFO level)
8. Ethernet link up (INFO level)
9. SLIC chip exception (WARNING and ERROR levels)
10. memory exception (ERROR level)

The Syslog uses USER facility. In addition to standard Syslog payload, it contains the following components: GS_LOG: [device MAC address][error code] error message

Example: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000] Ethernet link is up

Security

Download Configure

Downloads the current configuration of the GXW410X

Networks

Basic Settings

IP Address

There are two modes to operate the GXW410x:

DHCP mode: all the field values for the Static IP mode are not used (even though they are still saved in the Flash memory.) The GXW410x acquires its IP address from the first DHCP server it discovers from the LAN it is connected.

Using the PPPoE feature: set the PPPoE account settings. The GXW410x will establish a PPPoE session if any of the PPPoE fields is set.

Static IP mode: configure the IP address, Subnet Mask, Default Router IP address, DNS Server 1 (primary), DNS Server 2 (secondary) fields.

DHCP hostname

This option specifies the name of the client. This field is optional but may be required by some Internet Service Providers. Default is blank.

DHCP domain

This option specifies the domain name that client should use when resolving hostnames via the Domain Name System. Default is blank.

DHCP vendor class ID

Used by clients and servers to exchange vendor-specific information. Default is Grandstream GXW410x.

PPPoE account ID

PPPoE username. Necessary if ISP requires you to use a PPPoE (Point to Point Protocol over Ethernet) connection.

PPPoE password

PPPoE account password.

PPPoE Service Name

This field is optional. If your ISP uses a service name for the PPPoE connection, enter the service name here. Default is blank.

Preferred DNS server

This field will let the user enter a preferred DNS server to be used instead of the one acquired by the service provider.

Time Zone

Controls how the date/time is displayed according to the specified time zone.

Allow DHCP Option 2 to override Time Zone Settings

Default is No. If set to Yes, time zone settings will originate from the DHCP server.

Advanced Settings

Layer 3 QoS

This field defines the layer 3 QoS parameter which can be the value used for IP Precedence or Diff-Serv or MPLS.

Default value is 48. Its range lies from 0 to 63.

Layer 2 QoS

This contains the value used for layer 2 VLAN tag.

802.1q / VLAN tag: Default value is 0. Range lies from 0 to 4095.

802.1p Priority value: Default value is 0. Range lies from 0 to 7.

*** The above 2 settings need to be supported on the network and then configured accordingly on the GXW410x. Incorrect configuration will cause blocked access, which will result in Factory Reset as the only option to renew access.

Video Surveillance (HW version 1 only)

Default is No. Set to Yes, in order to enable the video in port. And configure the RTSP port number here(default port number 554).

Date & Time

NTP server

URI or IP address of the NTP (Network Time Protocol) server, which will be used by the phone to synchronize the date and time.

Allow DHCP Option 42 to override an NTP server

Default value is No. If set to Yes, the NTP server will originate from the DHCP server.

Self-Defined Time Zone (Yes/No)

Optional Rule:

This parameter controls whether the displayed time will be daylight savings time or not. If set to “Yes” and the Optional Rule is empty, then the displayed time will be 1 hour ahead of normal time. The “Automatic Daylight Saving Time Rule” shall have the following syntax: start-time;end-time;saving

Both start-time and end-time have the same syntax:

Month ,day ,weekday ,hour ,minute

month: 1,2,3,..,12 (for Jan, Feb, .., Dec)

day: [+|-]1,2,3,..,31

weekday: 1, 2, 3, .., 7 (for Mon, Tue, .., Sun), or 0 which means the daylight saving rule is not based on week days but based on the day of the month.

hour: hour (0-23),

minute: minute (0-59)

If “weekday” is 0, it means the date to start or end daylight saving is at exactly the given date. In that case, the “day” value must not be negative.

If “weekday” is not zero and “day” is positive, then the daylight saving starts on the first “day”th iteration of the weekday (1st Sunday, 3rd Tuesday etc). If “weekday” us not zero and “day” is negative, then the daylight saving starts on the last “day”th iteration of the weekday (last Sunday, 3rd last Tuesday etc). The saving is in the unit of minutes. The saving time may also be preceded by a negative (-) sign if subtraction is desired instead of addition. The default value for “Automatic Daylight Saving Time Rule” shall be set to

“03,11,0,02,00;11,04,0,02,00;60” which is the rule for US.

Examples:

US where daylight saving time is applicable: 03,11,0,02,00;11,4,0,02,00;60

This means the daylight saving time starts from 11^th March at 2AM and ends November 4th at 2AM. The saving is 60 minutes (1hour).

Settings

General settings

Use NAT IP

NAT IP address used in SIP/SDP message. Default is blank.

STUN Server

IP address or Domain name of the STUN (Simple Traversal of UDP through NATs) server.

Call Settings

G723 Rate

G723 encoding rate (6.3kbps or 5.3kbps)

Voice Frames per Tx

This field contains the number of voice frames to be transmitted in a single packet. When setting this value, the user should be aware of the requested packet time (used in SDP message) as a result of configuring this parameter. This parameter is associated with the first vocoder in the above vocoder Preference List or the actual used payload type negotiated between the 2 conversation parties at run time.

e.g., if the first vocoder is configured as G723 and the “Voice Frames per TX” is set to be 2, then the “ptime” value in the SDP message of an INVITE request will be 60ms because each G723 voice frame contains 30ms of audio. Similarly, if this field is set to be 2 and if the first vocoder chosen is G729 or G711 or G726, then the “ptime” value in the SDP message of an INVITE request will be 20ms.

If the configured voice frames per TX exceeds the maximum allowed value, the BudgeTone 200 will use and save the maximum allowed value for the corresponding first vocoder choice. The maximum value for PCM is 10(x10ms) frames; for G726, it is 20 (x10ms) frames; for G723, it is 32 (x30ms) frames; for G729/G728, 64 (x10ms) and 64 (x2.5ms) frames respectively.

Local RTP port

This parameter defines the local RTP-RTCP port pair the GXW410x will listen and transmit. It is the base RTP port for channel 0. When configured, channel 0 will use this port _value for RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and port_value+3 for its RTCP and so on. The default value is 5004.

RTP Loopback

Default value is No. If set to Yes, means no RTP if RTP streams between 2 internal ports.

Channel Settings

DTMF Method

This parameter specifies the mechanism to transmit DTMF digits. There are7 modes supported: in audio which means DTMF is combined in audio signal (not very reliable with low bit-rate codec), via RTP (RFC2833), or via SIP INFO. Multiple DTMF transmission schemas can be selected.

• 1 – in-audio
• 2 – RFC2833
• 3 – in-audio and RFC2833
• 4 – SIP Info
• 5 – in-audio and RFC2833
• 6 – SIP Info and RFC2833

1. 7 – in-audio, RFC2833, and SIP Info

No Key Entry Timeout

Default is 4 seconds.

Local SIP Listen Port

Default is ch1-8:5060++;. The ++ indicates increments by 2, so port 1 is set at 5060, port at 5062 and so on. This setting can be used with Round Robin and/or Flexible setting below to configure different ports to be placed under different Round Robin groups.

SRTP Mode

Default is disabled for all ports. The user can select to either enable it but not force it or force it on an individual port basis. When used the communication will be sent using Secure RTP.

Unconditional Call Forward to VOIP:

This is an extremely important setting to make sure incoming PSTN calls are picked up and forwarded to the correct VOIP destination.

User ID – This parameter allows users to configure a User ID or extension number to be automatically dialed upon FXO line off-hook.

SIP Server – You also need to specify the Profile of the user id configured above (p1 stands for Profile 1, p2 stands for Profile 2 and so on).

SIP Destination Port – Along with the user-id and Profile, you also have the option to choose the destination port where you would like to send the call. By default it should be set to ch1-x:5060; (x can be 4 or 8 depending on number of ports).

We can also specify a different destination for each port. For example under User ID we can type in: ch1:104;ch2:227;ch3-5:501;ch6,7:856.

Under Sip Server we can type in: ch1:p1;ch2-4:p2;ch5:p3

Under Sip Destination Port we can type in: ch1-2:5060;ch2:7080;ch3-8:5066++

T.38 Setting

This setting allows you to make several options related to facsimile.

You can select the method: T.38 or Pass through (G711)

You can select the fax transmission rates (2400/4800/7200/9600/12000/14400bps)

You can enable or disable ECM (Error Checking Mode)

Note: The user can only test the parameters for only one of the PSTN lines at the same time. In all cases please enter the telephone numbers as if the lines were to dial each other locally.

For the AC Impedance Test we only need to select the line to be tested by clicking on the AC impedance box corresponding to that line, telephone numbers are optional. Remember that the AC impedance test is usually used to reduce the echo that might be present in the line.

For the CPT test (call progress tones) we will test current disconnect as well. You will need 2 telephone numbers to perform the test. You can only perform the test on one line (row) at the same time and it will be the one that has the box checked for testing. This tested line will use another line connected to the gateway to perform the test by calling into it, this is why you will have to enter the telephone number for a second line to help with the test.

For CID detection you will need 2 telephone numbers to perform the test. You can only perform the test on one line (row) at the same time and it will be the one that has the box checked for testing. This tested line will use another line connected to the gateway to perform the test by calling into it, this is why you will have to enter the telephone number for a second line to help with the test.

To perform the test, please select the line you want to test and the desired test to be performed. Enter the information for this line as well as a second line if necessary. Then click on the update button and then reboot. Log back in and now you should see the information for the line selected as well as the check box already marked already there. Go ahead and start the test now, please wait a few minutes until the test is done.

Notes:

It is not required to enter a telephone number when testing for impedance, as the system does not place any actual calls for the test.

If you log into the Web Interface while the test is running will not interrupt the process.

Accounts

General Settings

Account Active

When set to Yes the SIP Profile is activated.

Account Name

A name to identify a Profile.

SIP Server

SIP Server’s IP address or Domain name provided by VoIP service provider.

Outbound Proxy

IP address or Domain name of Outbound Proxy, or Media Gateway, or Session Border Controller. Used by GXW410x for firewall or NAT penetration in different network environments. If symmetric NAT is detected, STUN will not work and ONLY outbound proxy can correct the problem.

Networks Settings

Use DNS SRV:

Default is No. If set to Yes the client will use DNS SRV to look up server.

NAT Traversal

This parameter defines whether the GXW410x NAT traversal mechanism will be activated or not. If activated (by choosing “Yes”) and a STUN server is also specified, then the GXW410x will behave according to the STUN client specification. Under this mode, the embedded STUN client inside the GXW410x will attempt to detect if and what type of firewall/NAT it is sitting behind through communication with the specified STUN server. If the detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the GXW410x will attempt to use its mapped public IP address and port in all of its SIP and SDP messages. If the NAT Traversal field is set to “Yes” with no specified STUN server, the GXW410x will periodically (every 20 seconds or so) send a blank UDP packet (with no payload data) to the SIP server to keep the “hole” on the NAT open.

Proxy-Require

SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.

Use OBP in Route

Utilizes outbound proxy in route.

SIP Settings

SIP Registration

This parameter controls whether the GXW410x needs to send REGISTER messages to the SIP Server. The default setting is “Yes”.

Unregister on Reboot

Default is No. If set to yes, the SIP user’s registration information will be cleared on reboot.

Register Expiration

This parameter allows the user to specify the time frequency (in minutes) for the GXW410x to refresh its registration with the specified registrar. The default interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes (about 45 days).

SIP Registration Failure Retry Wait Time

This parameter is mostly used by Service Providers. It prevents message REGISTER overload of SIP Server in case of downtime due to maintenance or power failure. By increasing interval length, common message load is decreased. Interval range is 1 – 3600 seconds.

SIP Transport

User can select UDP or TCP. Please make sure you’re SIP Server or network environment supports SIP over the selected transport method. Default is UDP.

Session Expiration

Grandstream implemented SIP Session Timer. The session timer extension enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or re-INVITE. Once the session interval expires, if there is no refresh via a UPDATE or re-INVITE message, the session will be terminated. Session Expiration is the time (in seconds) at which the session is considered timed out, if no successful session refresh transaction occurs beforehand. The default value is 180 seconds.

Min-SE

The minimum session expiration (in seconds). The default value is 90 seconds.

Caller Request Timer

If selecting “Yes” the phone will use session timer when it makes outbound calls if remote party supports session timer.

Callee Request Timer

If selecting “Yes” the phone will use session timer when it receives inbound calls with session timer request.

Force Timer

If selecting “Yes” the phone will use session timer even if the remote party does not support this feature. Selecting “No” will allow the phone to enable session timer only when the remote party support this feature. To turn off Session Timer, select “No” for Caller Request Timer, Callee Request Timer, and Force Timer.

UAC Specify Refresher

As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee or proxy server as the refresher.

UAS Specify Refresher

As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to use the phone as the refresher.

Force INVITE

Session Timer can be refreshed using INVITE method or UPDATE method. Select “Yes” to use INVITE method to refresh the session timer.

Enable 100rel

The use of the PRACK (Provisional Acknowledgment) method enables reliability to be offered to SIP provisional responses (1xx series). This is very important if PSTN inter-networking is to be supported. A user’s request to use reliable provisional responses is invoked by the 100rel tag which is appended to the value of the required header of initial signalling messages.

Refer-to uses Target Contact

Default is NO. If set to YES, then for Attended Transfer, the “Refer-To” header uses the transferred target’s Contact header information.  

INVITE Ring-no-answer Timeout

In case incoming call has arrived from PSTN to VoIP and INVITE message was generated by GXW device, the call will be disconnected after preconfigured timeout if not answered by VoIP extension.

Accept INVITE from Proxy Only

Default is YES. The device will authenticate and accept only incoming INVITE messages from the peered SIP server.

Audio Settings

Preferred Vocoder

The GXW410x supports up to 5 different Vocoder types including G.711 A-/U-law, GSM, G.723.1, G.729A/B. The user can configure Vocoders in a preference list that will be included with the same preference order in SDP message. The first Vocoder in this list can be entered by choosing the appropriate option in “Choice 1”. Similarly, the last Vocoder in this list can be entered by choosing the appropriate option in “Choice 8”.

Call Settings

User ID is Phone Number

If the GXW410x has an assigned PSTN telephone number, this field should be set to “Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will be attached to the “From” header in SIP request.

Early Dial

Default is No. Use only if proxy supports 484 response.

User Accounts

Note – The channels here are basically SIP endpoints that will act as clients registering to the SIP Server configured under the appropriate Accounts page.

Channels

It should be set same as the channel number (i.e 1, 2..4 or 8 depending on number of FXO ports). It is NOT the same as SIP Account ID.

SIP User ID

This is the SIP account information. Enter the SIP User ID part of the account.

Authentication ID

SIP service subscriber’s Authenticate ID used for authentication. It can be identical to, or different from SIP User ID.

Authen Password

SIP account password needs to be entered here.

Note: After entering the password, it will show up as blank but the password still remains active.

FXO Lines

Settings

Call Progress Tones

Using these settings, user can configure tone frequencies according to user preference. By default, the tones are set to North American frequencies. Frequencies should be configured with known values to avoid uncomfortable high pitch sounds. ON is the period of ringing (ON time in ms) while OFF is the period of silence. In order to set a continuous ring, OFF should be zero. Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern.

• “Dial tone”
• “Ringback tone”
• “Busy/Re-order tone”
• “Confirmation tone”

Please refer the document below to determine your local call progress tones

(http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf) or run the FXO Line Test (Table 9).

Tx to PSTN Audio Gain (dB)

Allows user to set a value in dB for transmission to PSTN Audio Gain. Default is 1. Range is from -12 to 12dB.

Rx from PSTN Audio Gain (dB)

Allows user to set a value in dB for receive from PSTN Audio Gain. Default is 0. Range is from -12 to 12dB.

Silence Suppression

This controls the silence suppression/VAD feature of G723 and G729. If set to “Yes”, when a silence is detected, small quantity of VAD packets (instead of audio packets) will be sent during the period of no talking. If set to “No”, this feature is disabled.

Echo Cancellation

When set to Y, Echo cancellation is enabled.

Enable Current Disconnect

When set to Y, Current Disconnect is enabled. Certain PSTN COs require this to be enabled, in order to realize disconnect signal from PSTN side. Default is Y.

If enabled use threshold: Default is 100ms. Range is 40ms to 800ms.

Certain PSTN service providers have a threshold time within which the line stabilizes after off-hook. It is entirely dependent on provider, however if you experience PSTN line detection issues, please modify this setting appropriately in 100ms increments.

If you are not sure if this option should be enabled please refer to Table 9 (FXO Lines Test Tab Definition). This tool will run an automated test to determine the proper PSTN configuration that the gateway should have to work with your Service Provider or analog PBX.

Enable Tone Disconnect

Default is No. If PSTN provider uses call progress tones then it should be set to Yes in order to realize disconnect tone. Please configure accurate Call Progress Tones on Channels webpage based on PSTN provider (or traditional PBX) settings.

If you are not sure if this option should be enabled or what Call Progress Tones are required please refer to Table 9 (FXO Lines Test Tab Definition). This tool will run an automated test to determine the proper PSTN configuration that the gateway should have to work with your Service Provider or analog PBX.

Enable Polarity Reversal

Default is No. This should be set to Yes only if the FXO lines are subscribed to PR service from PSTN Service provider. It is merely a PR detect feature.

***Note: If there is no PR service from provider on the FXO line, and this setting is configured to Yes, calls will not be successful.

Enable Call Answer Supervision

Default is No. If PSTN providers use the CAS then this option should be enabled. The Call Answer Supervision (CAS) is for billing—the telephone exchange and the customer need an accurate indication of calls through a network.

Silence Timeout

Terminate call after long silence detected. Default is 60 seconds, max 65536.

Incoming Call Timeout

Default value is 6 seconds. Incoming call will stop ringing when not picked up given a specific period of time.

AC Termination Impedance

Selects the impedance of the analog line connected to the FXO port on the GXW410x. Here is some basic information which may be helpful for initial configuration:

600 Ohm – North America;

270 Ohm + (750 Ohm || 150 nF) — Most of Europe

220 Ohm + (820 Ohm || 120 nF) – Australia, New Zealand

220 Ohm + (820 Ohm || 115 nF) – Austria, Bulgaria, Germany, Slovakia, South Africa

370 Ohm + (620 Ohm || 310 nF) – UK., India

If this parameter is not configured properly you may experience echo or static in the line. Please refer to Table 9 (FXO Lines Test Tab Definition). This tool will run an automated test to determine the correct impedance value to match your lines

Number of Rings Before Pickup

Default is 4. This is the number of rings the gateway will wait to send the call to the VOIP side in case the Caller ID has yet to be detected. If there’s CID information the call will be sent right away. If your lines don’t have the CID service set this to 1.

Caller ID Scheme

The GXW410x supports 5 different types of schemes:

1. Bellcore (US standard)
2. ETSI-FSK during ringing
3. ETSI-FSK prior to ringing with DTAS
4. ETSI-FSK prior to ringing with LR
5. ETSI-FSK prior to ringing with PR
6. ETSI-DTMF during ringing
7. ETSI-DTMF prior to ringing with DTAS
8. ETSI-DTMF prior to ringing with PR
9. ETSI-DTMF prior to ringing with PR
10. SIN 227 – BT
11. NTT (Japanese standard)

Please check with your PSTN service provider (or traditional PBX specs) for which caller ID scheme they/it support. If you are not sure about which to use please refer to Table 9 (FXO Lines Test Tab Definition). This tool will run an automated test to determine the proper Caller ID Scheme so the gateway can properly detect the Caller ID.

Similarly to the cases explained above we can specify a caller ID scheme for each channel independently.

Caller ID Transport type

Default is “relay via From header”. You may also select :

“relay via P_Asserted_Identity header”

“Disable” : Caller ID feature will be disabled.

“Send anonymous” : All calls forwarded to VOIP end will be sent as anonymous.

DIALING

Wait for Dial-tone

Default is Yes. When set to Yes, the gateway will recognize dial-tone from the Central Office (CO) before it completes call. If you can’t make an outbound call, set this is No.

Stage Method

Syntax – ch1-8:1; {all channels 1 to 8 are set to value 1 or 2}

Stage method can be set to either 1 or 2.

Set this parameter to 1 if you need to make a direct PSTN call from a VOIP endpoint. When you set it to 2, you will first dial one of the VOIP channel accounts from the VOIP endpoint, this will result in getting a PSTN line dial-tone to then dial out the destination PSTN number.

Most implementations require this setting to be configured to 1.

Min. Delay before Dial PSTN

Default is 500ms. This needs to be equal to or greater than the Current Disconnect threshold setting. Once the threshold is reached the gateway can dial out. This parameter should only be used if there are PSTN line detection issues.

Round Robin and/or Flexible

Default is rr:1-8;

The syntax is pretty straight-forward here. The rr stands for Round Robin and the numbers stand for the ports that belong to that round robin group.

For example:

rr:1-8; -> Round robin within the first 8 ports i.e. outgoing calls will be forwarded to the next available port within the group of ports 1 to 8.

rr:1,3-6,8;rr:2,7; -> Round robin within port 1,3,4,5,6 and 8; Second round robin group within ports 2 and 7 i.e. outgoing calls to ports 1,3,4,5,6 and 8 will be forwarded to the next available port within this group ONLY. Outgoing calls to port 2 and 7 will be forwarded to the next available port between ports 2 and 7 ONLY.

** In order to terminate a call on FXO port 2 or 7 you will need to change its Local SIP Listen port accordingly.

Prefix to specify Port

(1 stage dialing method)

Default is 99.

Syntax to USE this feature: prefix# (that is 99) + ch# (could be anything from 1 to 8) + dialing# will result in this call forwarded to FXO port (ch#) immediately.

Dial Plan

The Dial Plan feature implemented is applicable for VOIP to PSTN calls only. You may configure a dial plan based on the following grammar:

1. Accept Digits: 1,2,3,4,5,6,7,8,9,0,*,#,A,a,B,b,C,c,D,d
2. Grammar:

x – any digit from 0-9;

xx+ – at least 2 digit number;

xx. – at least 2 digit number;

^ – exclude;

[3-5] – any digit of 3, 4, or 5;

[147] – any digit 1, 4, or 7;

<2=011> – replace digit 2 with 011 when dialing

WARNING – illegal input will fall back to default

Example 1: {[369]11 | 1617xxxxxxx} – Allow 311, 611, 911, and any 10 digit numbers of leading digits 1617.

Example 2: {^1900x+ | <=1617>xxxxxxx} – Block any number of leading digits 1900 and add prefix 1617 for any dialed 7 digit numbers.

Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} – Allow any length of number with leading digit 2 and 10 digit-numbers of leading digit 1 and leading exchange number between 2 and 9;

If leading digit is 2, replace leading digit 2 with 011 before dialing

 Example 4: { [x#]+ | [x*]+ } – Allow any length of number with leading * or # in number to dial.

Default: PSTN Outgoing – {x+}

Note: If you do not plan to use this feature set to default {x+}

Hookflash Duration (X10ms)

Default 600ms. This value can accept any value in the 100-2000ms range.

Use DTMF Parameter from RFC2833 or SIP Info

Default Yes, No means to use DTMF parameter settings according to DTMF Digit Length, DTMF Digit Volume and DTMF Dial Pause.

DTMF Digit Length

Default value is 100ms. Please note that the value will be multiplied by 10ms

DTMF Digit Volume

Default value is -11dB.

DTMF Dial Pause

Default value is 100ms. Please note that the value will be multiplied by 10ms.

Line Analysis

Overview

Note: The user can only test the parameters for only one of the PSTN lines at the same time. In all cases please enter the telephone numbers as if the lines were to dial each other locally.

For the AC Impedance Test we only need to select the line to be tested by clicking on the AC impedance box corresponding to that line, telephone numbers are optional. Remember that the AC impedance test is usually used to reduce the echo that might be present in the line.

For the CPT test (call progress tones) we will test current disconnect as well. You will need 2 telephone numbers to perform the test. You can only perform the test on one line (row) at the same time and it will be the one that has the box checked for testing. This tested line will use another line connected to the gateway to perform the test by calling into it, this is why you will have to enter the telephone number for a second line to help with the test.

For CID detection you will need 2 telephone numbers to perform the test. You can only perform the test on one line (row) at the same time and it will be the one that has the box checked for testing. This tested line will use another line connected to the gateway to perform the test by calling into it, this is why you will have to enter the telephone number for a second line to help with the test.

To perform the test, please select the line you want to test and the desired test to be performed. Enter the information for this line as well as a second line if necessary. Then click on the update button and then reboot. Log back in and now you should see the information for the line selected as well as the check box already marked already there. Go ahead and start the test now, please wait a few minutes until the test is done.

Notes:

It is not required to enter a telephone number when testing for impedance, as the system does not place any actual calls for the test.

If you log into the Web Interface while the test is running will not interrupt the process.

Auto Detect

Line #

Enter the telephone (PSTN) number that corresponds to this line. Enter it as if you were going to dial it locally.

AC Impedance

Select this box if you want to test for impedance on the line that is on the same row as the checked box. Remember that you can only check one item at the same time.

CPT Detection

Select this box if you want to test for call progress tones and current disconnect threshold on the line that is on the same row as the checked box. Remember that you can only check one item at the same time.

External Number

Enter an external telephone (PSTN) number to be used as an auxiliary number for the test. This is used only if we do not have at least 2 PSTN lines connected to the gateway. This is only used for CPT and current disconnect threshold testing. In order to use this function you will have to monitor the Syslog output. This is only reserved for very advanced users.

External Call Timeout

This is the time the GXW will wait for the external telephone number to pick up during the test.

Apply test results automatically

Default is No. If selected on Yes, then all the results from the test will be applied automatically. If you select No you will have to monitor the Syslog output. This is only reserved for very advanced users.

Apply test results to all ports

Default is No. If selected on Yes, then all the test results will be applied to all ports on the gateway. If all the lines belong to the same service provider or PBX it will make sense to apply the results to all ports.

Error Timeout

This is the time the gateway will wait to exit the test mode, when something unexpected or an error has occurred.

Status

Hardware Revision

Hardware version number: Main Board, Interface Board

MAC Address

The device ID in HEX format. This is a very important ID for ISP troubleshooting.

IP Address

This field shows WAN IP address of GXW410x

Product Model

This field contains the product model info (GXW4104 or GXW4108)

Software Version

Program: This is the main software release. Boot and Loader are not changed often.

System Up Time

This field shows system up time since the last reboot.

Registered

This field indicates whether the different SIP Accounts configured under Channels page are successfully registered to the SIP server(s).

FXO Line Connected

This field will give the status of each physical FXO Line connected to the Gateway. It will update the status regularly.

Yes – Connected and Idle

Busy – Connected and Busy

No – Not connected

Additionally it will also provide real time Caller ID information of Incoming as well as Outgoing calls.

PPPoE Link Up

This field shows whether the PPPoE connection is running if connected to DSL modem.