Dynamix FXS-04, FXO-04, 2504, 2540 User Manual

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DYNAMIX

FXS-04 (

User Guide

2504) and

FXO

-04

(

2540

)

Contents

Dynamix

25xx ................................................................................... 3

Introduction ............................................................................................. 3

Dynamix

25xx Telephony Gateway ........................................................ 3

Physical Interface .................................................................................... 3

IP Network connection ............................................................................ 4

Environmental ......................................................................................... 4

Front Panel: LED Indicators ................................................................ 5

Rear Panel: .............................................................................................. 6

QUICK SETUP ....................................................................................... 7

Device Settings ....................................................................... 12

Network Configuration ......................................................................... 12

Device Advance Setting: ....................................................................... 15

NAT Setting .......................................................................................... 22

VOIP Advance: ..................................................................................... 33

Dialing Plan: ......................................................................................... 38

FXO Setting: ......................................................................................... 42

FXS Setting: .......................................................................................... 45

SIP Trunk: ............................................................................................. 50

Route Plan: ............................................................................................ 53

Appendix A --- Call Processing Flow (

FXO-04

) ...

..........

..................... 62

Appendix B --- System Recovery ......................................................... 63

Appendix C --- HTTP auto provisioning .............................................. 65

Dynamix 25xx

Introduction

This user manual contains both

FXO-04 (2540

) gateway. The word “

Dynamix FXS-04 (2504

Dynamix

25xx “ is going to use

) and

Dynamix

frequently to indicate both models common features. For specific feature to FXO or FXS gateway ONLY, we usually use the word “ FXO” or “FXS” to highlight their differences.

Dynamix 25xx Telephony Gateway

Dynamix 25xx is a 4 ports FXO (FXO-04) and 4 port FXS (FXS-04)

VoIP gateway which includes 1-WAN/1-LAN (management port) 10/100 base-T network environment. Field-proven quality of Voice communication and Fax transmission over IP broadband access network is to make Dynamix 25xx product to be an excellent solution for various VoIP applications.

Physical Interface

 Ethernet port (RJ-45, 10/100 base-T)

 1-WAN port, for connect to router, ADSL modem (ATU-R), or switch

hub directly.

 1-LAN port, for PC, management or other network devices

connecting.

 Telephony port (RJ-11)

 4-FXO ports, to connect to PSTN lines (FXO-04)  4-FXS ports, to connect to analog phone (FXS-04)

 Reset button (Factory Default)  DC power Jack  Status indicated LED

Indicates Power, Ethernet, Line, SIP and system status

IP Network connection

IPv4 (RFC 791)/IPV6 (RFC 2460) IPv6 Auto Configuration (RFC 4862) MAC Address (IEEE 802.3) Static IP DHCP Client (RFC 2131) PPPoE DNS Client TCP/UDP (RFC 793/768) RTP/RTCP (RFC 1889/1890) IPV4 ICMP (RFC 792)/IPV6 ICMP (RFC 4443) TFTP Client VOIP VLAN Support (802.1q/802.1p) HTTP/HTTPS Server QoS Support Support IPV4 only, IPV6 only or dual stack mode

Environmental

 Actual Dimension: 35 × 242 × 160 mm (Desktop)  Weight: 0.935kg (unit without packing)

 Operating Temp. & Humidity

- Temp.: 0℃～45℃ (32℉～113℉)

- Humidity: 10％～85% relative humidity, non-condensing

 Storage Temp. & Humidity

- Temp.: 0℃～55℃ (32℉～131℉)

- Humidity: 10％～95％ relative humidity, non-condensing

 AC to DC Power Adaptor:

- INPUT: AC100V-240V, 50/60Hz

- OUTPUT: DC 12V, 1.5A

 Regulatory Compliance: FCC (Part 15, Class B) & CE

Front Panel: LED Indicators

Dynamix FXO-04, FXS-04

LED Description

Power When the power adapter is connected, the LED will light

up green.

Status When system startup successfully, the LED will light up

green.

Proxy When the gateway is registered successfully to a SIP

Proxy, this will light up green.

WAN This LEDlights up green when the gateway’s WAN port is

physically connected to the public internet. When data is transmitted through this port, it will flash green.

The default IP of WAN port is 10.1.1.3.

LAN This LEDlights up green when the gateway’s LAN port is

physically connected to a local network (Refer to Rear Panel section). When data is transmitted through this port, it will flash green.

The default IP of LAN port is 192.168.123.123.

Port1~Port4 The status LED for FXO and FXS port 1-4, these LED light

up amber orange when connected phone is engaged in a conversation mode (FXO). It will flash amber orange when there is an incoming call (FXS).

Rear Panel:

when the connected phone is engaged in a conversation.

Dynamix FXO-04, FXS-04

Item Description

Reset Press and hold over 5 seconds to reload factory default

setting, this will erase all existing settings configured on

this gateway. L1-L4 (FXO Gateway Only)

T1-T4 (FXS Gateway Only)

LAN 10/100 Base-T RJ-45 socket for LAN port , connects to PC

WAN 10/100 Base-T RJ-45 socket for WAN port, connects to

DC 12V The power socket, input AC 100V~240V; output DC12V,

The status LED for FXO port 1-4. When there is no PSTN

line connected, this LED will become blinking to remind

you. When PSTN line is connected and no talking, the LED

will be off. When a line is using, the LED will become

steady light up. ( FXO-04 only)

The status LED for FXS port 1-4, this will light up amber

orange when the connected phone’s handset is lifted, or

It will flash amber orange when there is an incoming call.

( FXS-04 only)

for management purpose.

wide area network.

1.5A

QUICK SETUP

Setp1: Setup the administrative PC’s IP address to be same as 25xx and

connect the Ethernet cable into WAN or LAN port. Start IE6.0 (or later version)

to navigate 25xx web management system by typing the default URL

which is

(through WAN port). The screen will display User Name and Password (the default user id is Note : Dynamix 25xx Web browser does not support FireFox Web browser. DO NOT use FireFox to configure Dynamix 25xx, otherwise you may meet an unstable configuration.

http://192.168.123.123 (through LAN port) or http://10.1.1.3

root and user password is root). (Fig.1)

Fig.1

Step 2: After login, the screen shows the Home page of Dynamix 25xx. (Fig.2)

Note: The Web Page may differ at FXS or FXO Setting due to the model you are using.

Fig.2

Change Default IP Network:

Step 3: After successfully logon to the system, we need to change the network configuration. Click Device Setting > Network to setup the service network interface (WAN) parameters. Enter the desired IP address, Subnet mask and default gateway or select “DHCP” or “PPPOE”. Apply the change by clicking Apply button as fig (Fig.3).

Note: If your WAN port are setting in the 10.x.x.x segment IP address, please make sure that you also need to change the LAN port to other segment such as

192.168.x.x

(Fig.3)

Change Default Time setting:

Step 4: When re-logon to the new IP address, the next step is to setup the system time zone. Click Device Setting > Time to setup the system. Enter the current SNTP server IP address, time zone and daylight saving parameters. Apply the change by clicking

Apply button. (Fig.4)

(Fig.4)

Modify SIP Account Parameter:

Step 5: The next step is to add a SIP trunk for VOIP calling. FXO-04 (2540), it is necessary to assign one SIP trunk for VOIP calling. However, to assign one SIP trunk to FXS-04 (2504) is optional. Click “SIP trunk and new” to

create

the required sip trunk. Enter the trunk ID to 1 and input those SIP parameters. Apply the change by clicking Apply button. (Fig.5)

(Fig.5)

Modify FXS SIP Settings: ( FXS-04 only)

Step 6: Set the SIP proxy server IP address for FXS calling. For DYNAMIX FXS-04, all FXS ports are using the same SIP proxy setting. If you need to use different SIP proxy server, please use SIP trunk instead. Click FXS Settings > SIP Proxy to set the dedicated FXS SIP proxy server.

(Fig. 6) Step 7: Setup each FXS line’s parameters by clicking the line ID from FXS settings > FXS Line. Modify the SIP register information and apply it. (Fig. 7)

(Fig. 7)

Soft Reset Dynamix 25xx:

Step 8: After modify basic setting. It is required to reset Dynamix 25xx. Click

Maintenance > Maintenance > Soft-Reset or Reboot to take effect.

Apply the change by clicking Apply button. (Fig.8)

(Fig.8)

Check Dynamix 25xx Registration Status:

Step 9: After soft-reset or reboot. >Click Status > SIP Trunk Status to check whether registration was successful or not. (Fig.9.1)

(Fig.9.1)

>Click line status to check registration status. (Fig.9.2)

(Fig.9.2)

Through the above settings, the Dynamix 25xx should be able to do the following features: For FXO: (2 stage dialing use)

1. For PSTN incoming call, the caller will hear a dial tone. Then the caller can

dial a VOIP number and it will use the setup SIP Trunk 1 to make SIP call out.

2. For VOIP incoming call, FXO-04 will off hook a FXO port and dial to

PSTN Phone.

For FXS:

1. The user can pick up the handset and hear dial tone. Call out and talk.

2. For VOIP incoming call to a dedicated FXS number, this phone will ring and can answer to talk.

Device Settings

From this setting category, all devices related to parameters can be found here.

Network Configuration

> Network

Parameter Description:

Setting:

 IP Support: IP stack to be supported (IPV6 and IPV4 or IPV6 or IPV4

only)

WAN Setting:

 Network Type: support “Fixed IP”;”DHCP”;”PPPOE”  IP Address  Netmask: IPV4 network subnet mask  Default Gateway: IPV4 Default gateway  IPV6 Network Type: Auto configuration or manual configuration

IPV4 address

 IPV6 IP Address: IPV6 address  IPV6 IP Gateway: IPV6 default Gateway  IPV6 IP Prefix Length: IPV6 prefix length  DNS Server1: Primary DNS Server IP network  DNS Server2: Secondary DNS Server IP network  VOIP VLAN: Enable VOIP VLAN or not. When enable VOIP VLAN, the

WAN port can be only accessed by VLAN. If it is required to manage the

Dynamix 25xx, Administrator can use LAN port to access this gateway

instead.

 VOIP VLAN ID(2-4096): VLAN ID Used can be used range.

Note: the default WAN IP address is 10.1.1.3.

LAN Setting:

Voice gateway mode:

used for register to SIP Server or data/voice routing.

NAT mode: DHCP function on the LAN port. The LAN port functions as a DHCP server, network devices connected to them will be assigned one IP addresses according to DHCP server IP range. (Please refer to command “NAT setting” on the left side commands how to define DHCP IP address.)

 IP Address

using 10.x.x.x IP segment).

 Netmask: IPV4 network subnet mask  IPV6 IP Address: IPV6 address  IPV6 IP Prefix Length: IPV6 prefix length

Bridge mode: At this mode, both WAN and LAN ports are configured to Switch/Hub features. LAN port access to WAN port directly.

This LAN port is used for management purpose, not

IPV4 address (please set to 192.168.x.x if your WAN port is

Note: default LAN IP address is 192.168.123.123

DDNS (DynDNS) Setting: DDNS (DynDNS): enable or disable dynamic DNS feature. Domain Name: input your Domain Name User Name: input your user name Password: input your password

Device Time Setting:

Dynamix

Device Setting > Time

25xx support SNTP with time zone and daylight saving.

Parameter Description:

 Current Time: Current Time, date and year display.  NTP Time Server  NTP Refresh Interval(sec)

seconds

 Time Zone:

SNTP time server

The interval time to sync NTP server in

The time-zone where Dynamix 25xx is located.

IP address

- Standard: Use a predefined standard time zone

- Customize: Use a user defined time zone

 Daylight Saving: Auto adjust daylight saving timer or not  Daylight Bias: The offset added to the Bias when the time zone is in

daylight saving time

 Daylight Start: The date that a time zone enters daylight time

- Month: 01 to 12

- Week Day: Sunday to Saturday

- Apply Week (Day:01 to 05, Specifies the occurrence of day in the

month; 01 = First occurrence of day, 02 = Second occurrence of day, ...and 05 = Last occurrence of day)

- Hour: 00 to 23

 Standard Start: The date that a time zone enters daylight time

- Month: 01 to 12

- Week Day: Sunday to Saturday

- Apply Week (Day:01 to 05, Specifies the occurrence of day in the

month; 01 = First occurrence of day, 02 = Second occurrence of day, ...and 05 = Last occurrence of day)

- Hour: 00 to 23

Device Advance Setting:

> Advance

Parameter Description:

 HTTP Port: The Administrator Web service port (the default is 80)  HTTPS Port: The https web service port (the default is 443)  Telnet Port: The telnet service port (the default is 23), when click the

disable option; the Telnet service will be rejected.

 HTTP/HTTPS Service Access on WAN: When click the disable option;

the WEB service will be rejected on WAN port, so please be careful with this function. If you wanted to enable WAN port again, you need to access this device from its LAN port to connect to WEB pages and enable WAN port.

User Login Setting:

Three level of users can be used, administrator, supervisor, user. Each level of users has different predefined access level.

>User Login

Parameter Description:

 Administrator: The administrator level user who has full access

authority to Dynamix 25xx.

 Supervisor: The supervisor level user who has limited administrative

access right.

 User: The user access right which only allows to set some user related

features.

 User ID: Login User ID  Password: Login Password  Confirm Password: Confirm new password again  Language: The desired web page language used when the account was

Debug Settings:

Dynamix 25xx provides the real time debug to syslog or through telnet

interface. It generates the debug information based on debug level and modules. Since the generating debug will consume system resource, it is recommended to turn on only for necessary and under Dynamix FAE’s instruction.

Debug

Parameter Description:

 Syslog: Enable or disable to send system information to syslog server or

not

 Check for start from Any Time: Always Send: Always send syslog or

only during a specified time range.

 Syslog Start (YYYY/MM/DD HH:MM): Always Send: Always send

syslog or only during a specified time range.

 Syslog Stop (YYYY/MM/DD HH:MM): The syslog stop sending time.  SyslogD Server: SyslogD server IP address  Syslog Port: syslog server service port (default is 514)  DSP Debug: Enable or disable to send DSP information to capture log  DSP Capture server: syslog capture server IP address  DSP Capture port: syslog capture server service port (default is 50000)

> Event Notice

Dynamix 25xx can send Syslog Event Notice when it had the following cases:

1. Register Failure or re-registered

2. FXO is plug or unplug

3. Ethernet reconnected

4. System started

 Syslog: Enable or disable to send system event to syslog server or not  Syslog Port: syslog server service port (default is 514)

Auto Provision:

The Dynamix 25xx can be provisioned by HTTP Server for large deployment. Please contact Welltech for availabilities.

>Provisioning

Select 9510: ( This feature is not yet available now. Please don’t select at present )

Parameter Description:

(This function is available yet for WellEMS 9510)

 Provisioning Service: Enable auto provisioning service by WellEMS 9510

or not.

- Enable: Enable the service and use manual configured EMS server

parameters.

- Disable: Disable the auto provisioning service.

- Discovery: To automatically discover the EMS server or not. By using

this mode, WellEMS 9510 need to be the same IP network in order to make the IP broadcasting work.

 EMS Discovery Port: WellEMS 9510 service auto discovery broadcasting

port (default is 61005).

 EMS summary refresh interval: How long the 25xx will report its

summary status to WellEMS 9510 in seconds.

 EMS IP address: The WellEMS 9510 server IP address  EMS Server Port: The WellEMS 9510 Server port

Select Http: This feature is ready to use now.

 Http Configuration: Enable or Disable  Http Config URL: internal used only  Refresh interval(minute): interval to check whether have a new

configuration/firmware or not in minutes

 User ID: specify the login id for http authentication  Password: specify the password for http authentication

>SNMP

SNMP Agent:

 SNMP Agent: Enable SNMP or not.  Read Only Community Name: The community name to read through

SNMP protocol

 Read Write Community Name: The community name to read and

write through SNMP protocol.

 SNMP Agent Access on WAN: Enable SNMP to be accessed through

WAN port or not.

Trusted Peer:

 Type:

 Any Address: Any address can retrieve the SNMP information.  Specify an IP Address: Only the IP address listed can retrieve

the SNMP information. Normally, it will be the SNMP manager IP address.

 Specify a Subnet: Only the network specified can retrieve the

SNMP information.

 IP address: The IP address for a trusted peer  Subnet Mask: The network mask for a trusted peer

SNMP Trap:

 SNMP Trap: Enable SNMP trap or not  Destination: The IP address for SNMP manager to receive the SNMP

trap

 Community: The communication name for sending the SNMP trap

NAT Setting

The Dyanmix 25xx can support NAT, 2 ethernet leg (gateway mode) or bridge mode. Here is the setting for NAT related service.

> DHCP Ser. (DHCP server)

 DHCP Server: Enable DHCP server or not.  DHCP Client Range(Start IP): specify DHCP client lease start IP  DHCP Client Range(End IP): specify DHCP client lease end IP  DHCP Route Opt: specify the default gateway  DHCP Subnet Opt: specify the subnet mask.  DHCP DNS1 Opt: specify the DNS server  DHCP DNS2 Opt: specify the DNS server

> UPNP (universal plug and play server)

 UPNP IGD: Enable UPNP server or not.

> Bandwidth (Bandwidth Control)

By using bandwidth control feature, the user can manage the traffic based on their needs.

Bandwidth Control:

 Bandwidth Control: enable bandwidth control or not.  Download Bandwidth: specify total bandwidth for download (unit:

kbps). 0 indicates no limitation.

 Upload Bandwidth: specify total bandwidth for upload (unit: kbps). 0

indicates no limitation.

Maximum Bandwidth and Reserved Bandwidth:

 Setup Method: bandwidth control method, percentage or specify the

required bandwidth

 percentage : total bandwidth

 priority 1: highest priority percentage  priority 2: Normal priority percentage  priority 3: low priority percentage

 specific :

 priority 1 – Download: highest priority download bandwidth  priority 2 – Download: normal priority download bandwidth  priority 3 – Download: low priority download bandwidth

 priority 1 – Upload: highest priority upload bandwidth  priority 2 – Upload: normal priority upload bandwidth  priority 3 – Upload: low priority upload bandwidth

In order to set which target is belonged to which priority. The following is the setting method for target’s priority.

IP Target

 Priority: Priority value for the target  Type: The target type is set to IP  Configure Type: unique IP or a range of IP addresses

 Unique:

 IP Address: the IP address to be set

 IP Range:

 Start IP: The starting IP for a range  End IP: The stopping IP for a range

Port Target

 Priority: Priority value for the target  Type: The target type is set to port number  Configure Type: unique port number or a range of port number

 Unique:

 Port: the port number to be added  Protocol: protocol for the port

 Port Range:

 Start port: the starting port number  End port: the stop port number  Protocol: protocol for the port range

Application Target

 Priority: Priority value for the target  Type: Application  Application: the list for the application

DSCP target

 Priority: Priority value for the target  Type: DSCP value  DSCP: The DSCP will be mapped to the priority

The Dynamix 25xx support firewall features as below.

> URL Filter

 URL Filter: the specified url will be blocked

> IP Filter

 IP Filter: The specified IP address to be blocked

 MAC Filter

 MAC Filter: The MAC address to be blocked

> APP Filter

 APP Filter: application to be blocked

 Port Filter

 Port Filter: port number or range to be blocked

> Port Fwd

The wellgate 25xx support port forward feature as below

 Port Fwd: enable port forward feature or not  Port Range: Starting and stopping port to be forwarded. If you are

using only 1 port, please set the starting port equal to stopping port.

 Protocol: TCP, UDP or both are used for port forward  Local IP address: The LAN side IP address to be forwarded  Local Port: The LAN side port to be forwarded. If you are using the port

range, this port indicates the starting port.

VOIP Parameters Setting

SIP Parameters:

VOIP Setting

> SIP

Parameter Description:

 Session Timer: Enable session timer or not (RFC 4028)  Session Expires (sec): This is the setting of initial session timer

expires time according to rfc4028 - Session Timers in the Session Initiation Protocol.

Min SE: The minimum session timer allowed when receiving a call with session timer value according to RFC 4028.

 Session Timer Refresh Method: The session timer refresh method  PRACK: Enable provision ACK or not (RFC 3262)

- None: Disable PARCK

- Supported: When select this mode, 100rel will be added to the

support list. It indicates 25xx can support the PRACK but not mandatory.

- Require: PRACK is mandatory required.

 SIP Local Port: The SIP local service port (default is 8080)  SIP Qos Type: Quality of Service Type for SIP signaling

- None: Not using QOS Tag and not enables QOS.

- DiffServ: Differentiated Services Value. Input DSCP value 0-63 for

DSCP

- TOS: Type of Service which include IP precedence value and TOS.

 Accept Proxy Only: Only accept the call coming from the SIP proxy.

Not accept peer to peer call.

> Audio

 Codec 1~5: The preference codec priority

 G.711u Payload Size: G.711 u-Law payload size  G.711a Payload Size: G.711 A-law payload size  G.729 Payload Size: G.729A payload size  G.723.1 Payload Size: G.723.1 payload size  Bit Rate: G.723.1 bit rate used

 5.3K bit rate is used  6.3K bit rate is used

 Codec Priority: Selection order to match the remote SDP for codec

selection.

 Local SDP Order: Use local SDP order to match codec  Remote SDP Order: Use Remote SDP order to match codec

 DTMF Relay:

In-Band DTMF: use inband DTMF instead of out of band.

RFC 2833(fall back to SIP-INFO): Use RFC 2833 if the SDP

negotiation could be done. Or use SIP INFO for DTMF relay.

SIP INFO: Use SIP-INFO DTMF relay

RFC 2833(fall back to Inband): Use RFC 2833 if the SDP

negotiation could be done. Or use inband DTMF transmission.

 Silence Suppression:

Enable: Start the voice activity (silence) detection and send SID

when detect silence.

Disable: Send silence packet as normal voice packet (no silence

detection)

 RTP Basic Port: The RTP starting port. Each channel will be added

additional 10. For example, the RTP basic port is 16384, thus call 1 will use 16384 while call 2 will use 16394 etc.

 RTP Qos Type: IP QoS tag for RTP stream

 DiffServ: The differentiated service QoS tag will be used.

Input DSCP value 0-63 for DSCP.

 TOS: Type of Service which include IP precedence value and

TOS.

> Tone

The setting page is used to setup the tone to be generated or detected. The detected tone is the Disconnect 1 & 2 (for FXO use) and the others are for generating (when FXS received the “bye” from IP side or waiting time

out by analog phone which keeps handset pick up, it will send busy tone to analog phone). The disconnect tone is very important for PSTN

status supervision to release FXO port after call was dropped.

Please use Country Template to select the country profile which will be applied. Click Use to load those country tone parameters to system and change if necessary. For those countries are not showed in the list, please select a

closed country and edit to match your country. You can send an email with the tone definition to Welltech if you would like to put your country tone into the list.

> NAT Traversal

The Dynamix 25xx support the following NAT traversal methods

 NAT Traversal:

 Disable: Disable NAT traversal features  STUN (Type 1,2): Enable STUN for NAT traversal. Since

 STUN (All): No matter which NAT type server are used,

 UPNP: Enable UPnP client for NAT traversal. Please note that  Behind NAT: Use DMZ for NAT traversal

STUN can be used only for type 1 and type 2 NAT server, it is recommended to use this option. When STUN client detect the used NAT is type 3 NAT, it will stop the STUN feature.

 STUN Server: STUN Server IP address

STUN is always to be used for NAT traversal.

 STUN Server: STUN Server IP address

the IP sharing box need to support uPnP feature.

 IP Sharing Address: public IP sharing address. You

need to specify the port mapping or DMZ for all required ports.

VOIP Advance:

SIP >

Parameter Description:

 SIP Hold Type: SIP on hold message sending method.

- Send Only: Set the SDP media to send only when send an on-hold

SIP message.

- 0.0.0.0: Set the SDP connection to 0.0.0.0 when send an on-hold

SIP message.

- Inactive: Set the SDP media to inactive when send an on-hold SIP

message.

 SIP Compact Form: Enable SIP compact form or not. When enable this

feature, the connected SIP proxy is required to support compact form.

 Session Refresher: Who will send dialog to keep alive message

(re-invite or update).

- UAC: User Agent Client will do the refresh (default setting)

- UAS: User Agent Server will do the refresh

 SIP T1 (msec): T1 determines several timers as defined in RFC3261.

For example, when an unreliable transport protocol is used, a Client Invite transaction retransmits requests at an interval that start at T1 seconds and doubles after every retransmission. A Client General transaction retransmits requests at an interval that starts at T1 and doubles until it reaches T2. (Default Value: 500ms) **

 SIP T2 (msec): Determines the maximum retransmission interval as

defined in RFC3261. For example, when an unreliable transport protocol is used, general requests are retransmitted at an interval which starts at T1 and doubles until reaches T2. If a provisional response is received, retransmission continue but at an interval of T2. (Default Value: 4000ms) **

 SIP T4 (msec): T4 represents the amount of time the network takes to

clear message between client and server transactions as defined in RFC3261. For example, when working with an unreliable transport protocol, T4 determines the time that UAS waits after receiving an ACK message and before terminating the transaction. (Default Value: 5000)

 Invite Linger Timer: After sending an ACK for an INVITE final response,

a client cannot be sure that the server has received the ACK message. The client should be able to retransmit the ACK upon receiving retransmissions of the final response for this timer. This timer is also used when a 2xx response is sent for an incoming Invite. In this case, the ACK is not part of the Invite transaction.

 General Linger Timer: After a UAS sends a final response, the UAS

cannot be sure that the client has received the response message. The UAS should be able to retransmit the response upon receiving retransmissions of the request based on this timer.

 Cancel General No Response Time (msec): When sending a CANCEL

request on a General transaction, the User Agent waits cancel General No Response Timer milliseconds before timeout termination if there is no response for the cancelled transaction(Default Value: 10000ms).**

 General Request Timeout Timer (msec): After sending a General

request, the User Agent waits for a final response general Request Timeout Timer milliseconds before timeout termination (in this time the User Agent retransmits the request every T1, 2*T1,…T2,…milliseconds)**

 Cancel Invite No Response Timer (msec): When sending a

CANCEL request on an Invite request, the User Agent waits this timer before timeout termination if there is no response for the cancelled transaction.

 Provisional Timer (msec): The provisionalTimer is set when receiving

a provisional response on an INVITE transaction. The transaction will stop retransmissions of the INVITE request and will wait for a final response until the provisionTimer was expired. If you set the provisionTimer to 0, no timer is set. The INVITE transaction will wait indefinitely for the final response.

 First Response Timer (msec): When sending a request out, the User

Agent waits this timer for any response received from UAS. If timer is expired and no any SIP message is received, the User Agent will think the request is failed. The default is 5 seconds.

 MWI Subscript Expires (sec): You can Enable or Disable the MWI

subscribe. The default is 600 sec. If a new voice mail is arrived, the stutter tone will be used instead of regular dial tone. This feature is dedicated to FXS only.

 Line Congestion Code: when callee's end system was contacted

successfully but the callee is busy and does not wish to take the call at this time, the system will response the code, default is 600. (FXO use)

 SIP-Info Flash Mode: when you enable the feature, system will make

flash key to send sip message by sip-info.

 Encrypt:

 Disable: disable encryption function.  Welltech encryption: Enable Welltech proprietary encryption for

SIP signaling and RTP. It is required a Welltech SIP proxy server (WS6500 or SIPPBX 6200) to work with this feature. When enable it, you can hide your VOIP traffic from ISP’s monitor.

 External encryption: for custom encryption, it is valid now, if you

want add the function to mach your proxy, please contact with Welltech’s sales.

Audio>

The setting page includes the device related audio settings.

 RFC 2833 Payload Type: 96 or 101. It is recommended to use 101.  DTMF Send On Time(msec): When generate DTMF, the DTMF ON

time will be sent (default value is 70 ms)

 DTMF Send Off Time(msec): When generate DTMF, the DTMF OFF

time will be sent (default value is 70 ms)

 DTMF Detect Min on Time (msec): The minimum DTMF ON time

period will be processed as a regular DTMF event. A smaller ON time than this will be ignored. The default value is 60ms.

 DTMF Detect Min off Time (msec): The minimum DTMF OFF time

for the same DTMF value. A smaller OFF time than this and the new DTMF digit is the same as previous one will be handled as 1 digit only(same digit and not a new digit).

 DTMF Relay Volume: The DTMF relay volume  T.38 Fax Volume: The T.38 fax relay volume  T.38 Redundant Depth: The T.38 redundant packet depth. It could 0

(no redundant), 1 or 2. It is recommended to set to 2.

 T.38 ECM: The t.38 error correction mode. Default value is ON.  Min Jitter Buffer (msec): The minimum delay time of Jitter buffer.  Max Jitter Buffer (msec): The Maximum delay time of Jitter buffer.  Max Echo Tail Length (G.168): Enable the echo cancellation feature.

The default setting is “128ms”.

 Jitter Opt. Factor: Jitter buffer dynamic factor for optimize. Please

set to 7 unless under Dynamix’s instruction to change.

Ring>

The ring cadence, voltage and frequency were configured to the phone.

 Frequency (10~70HZ): Specify the ringing frequency value

(default is 20HZ)

 Ring on (0~8000ms): Specify the ringing on value (default

is 1000msec)

 Ring off (0~8000ms): Specify the ringing off value (default

is 2000msec)

 Ring level (10~95volt): Specify the ringing level (default is

94 volt)

Dialing Plan:

General>

 First Digit Time Out:

receiver is off hook. The range is 1~60 sec.

 Inter Digit Time Out: Specify the interval of input digits, if the

interval is over the setting, the system will end up the dial and send out

the DTMF. The limitation

 End of Digit: The assigned key will be treated as end of dial.  Retrieve Number: it will forced the line to retrieve, if 25xx makes

transfer to other devices but the devices DO NOT answer and go into voice mail service. You can press the preprogram code to retrieve the line. Default code is “*#”.

Dialing Rule>

Specify the duration of dial waiting when the

range is 1~60 sec.

Dialing rule is used to speed up the dialing procedure. Some user don’t like to use the end of dialing digit such as “#”, the administrator can use dialing rule instead. The longest prefix will be matched first.

 Dialed Prefix: The prefix to be matched  Max Digits: The digits will be received based on the Dialed Prefix.

The following is an example for dialing rule: Mobile call is starting with 09 and it is 10 digits Long distance call is starting with 0 and it is 10 digits International call is starting with 00 and its max digit should be less than 32

The others are local call and 8 digits

Emergency call is starting with 1 and 3 digits

The Dialing rule can be set as follows: Prefix, max digits 09, 10 0, 10 00, 15 1, 3 2, 8 3, 8 4, 8 5, 8 6, 8 7, 8 8, 8 9, 8

Digit Manipulation>

The Digit Manipulation (DM) will be processed based on prefix and DM group after the DNIS is determined.

 DM Group: Different DM group have different case to be used.

 FXO: This DM group is used for FXO 2 stages dialing. After the

DNIS (Called party messages) is collected, this DM group will be processed before enter the routing procedure.

 VOIP: This DM group is used for VOIP incoming call. After the

DNIS is collected in 2 stages dialing or 1 stage dialing DNIS, this DM group will be processed before entering the routing procedure.

 1-4: These DM groups are used for backup routing purpose.

When a backup routing is used, the administrator can select a DM group to be processed before starting the backup route.

 Matched Prefix: The prefix to be matched for DM. The longest prefix

will be matched first.

 Matched Length: Set to 0 to ignore the length. The other 1-32 are the

digit length to be matched as a condition.

 Start Pos: The start digit position to be replaced.  Stop Pos: The stop digit position to be replaced.  Replace Value: The value to be replaced.

Example of Digit Manipulation Settings:

Prefix Len Start

Pos

886 0 0 0 002 8862123456 0028862123456 886 12 0 0 002 8862123456 8862123456 886 0 2 5 002 8862123456 8800223456 886 0 30 30 002 8862123456 8862123456002 886 0 1 6 8862123456 83456

Stop Pos

Replace Value

Test DNIS Result DNIS

Phone Book>

Phone Book is used to peer to peer call.

 Name: Just support called number, if you input words on here, it will

routes to proxy server.

 Tel No: input called number and IP address, please following this sample

of picture, as the format of “number@uri:port”. (default port is 5060)

 Export: To backup the phone book records.  Import: To reload setting of phone book.

FXO Setting:

The FXO Setting contains the FXO related parameters.

 Line ID: FXO line (L1 to L4)  State: The line is active or not  TEL No: The reference telephone number (e.g. PSTN TEL of line)  Hotline TEL:

FXO line>

If hot line is set, this field shows the hot line number.

 User ID: FXO Line number (L1 to L4)  User Type: the line type which is FXO  Line State: Set to active if you would like to use this line. Otherwise,

set to Inactive.

 TEL NO: This field can be used as a reference remark for this line.

Normally, you can put the connected PSTN line’s phone number here for reference.

 Polarity Reversal Detection: When enable the Polarity Reversal

Detection; the Dynamix FXO-04 will use the polarity reversal signal as call was established for FXO outgoing call and start to count talking time for Billing purpose. When disable the polarity Reversal Detection, the Dynamix FXO-04 will use “

to set time (seconds) to answer the SIP call.

 Current Drop for disconnect: Use Line current drop as a disconnect

supervision to release FXO port.

 Incoming call handling: The call handling policy for a FXO incoming

call.

 Hot line TEL: When a FXO incoming call was detected and

after the PSTN Answer Ring Count, Dynamix FXO-04 will send the SIP call to the specified hot line TEL number through the Route Plan.

 2 Stage Dialing: When a FXO incoming call was detected and

after the PSTN Answer Ring Count, Dynamix FXO-04 will answer this call and play either dial tone or voice greeting file for 2 stages dialing to VOIP SIP Trunk.

 Playback voice file: To enable playing voice greeting file or not. (Only

FXO use )

 Repeat Count: Repeat how many counts to Play voice greeting file.

( Only FXO use )

 Voice file name (MuLaw-mono 8K): Specify the file path and file

name to upload. Please make sure that the file format needs to be G.711U, 8K, 8 bits raw file. (Only FXO use )

 Flash Time: Flash Time will be sent to PSTN line (internal use only)  FAX Relay: Enable T.38 Fax Relay or T.30 Fax Bypass or not.

 Input(Encode)Gain: Adjust the volume from PSTN to VOIP (default

is 0 db)

 Output(Decode)Gain: Adjust the volume from VOIP to PSTN (default

is 0 db)

 Dialing Answer Delay Time (sec): When the polarity reversal

detection is disabled, Dynamix FXO-04 will answer the call(establish call between VoIP and FXO) after time out for Billing application purpose. After the DTMF digits dialing, Dynamix FXO-04 will send 183 with SDP to SIP Trunk to enable the voice path for VOIP side.

 PSTN Answer Ring Count: This ring count is used for called ID

detection and 2 stage dialing.

(T.30 Fax Bypass only supports G711a law)

 If the caller ID is sending during the first ring and second

ring, this parameter should be set to greater than or

Dialing Answer Delay Time” command

equal to 2.

 If the caller ID is sending before the first ring, this

parameter can be set to greater or equal to 1. After the ring count was reached, 25xx will answer the call and play voice greeting file if 2 stage dialing is selected. Or, make the VOIP call out directly if hot line mode and number is selected.

 Caller ID Mode: The detected Caller ID specification for the line based

on selected country list or FSK or DTMF.

FXS Setting:

The FXS line setting includes each line number and SIP proxy settings.

 Line ID: FXS line (T1 to T4)  State: The line is active or not  TEL No: The telephone number  Hotline TEL:

FXS line>

If hot line is set, this field shows the hot line number.

 User ID: FXS Line number (T1 to T4)  User Type: The line type, FXO or FXS  Line State: Set to active if you would like to use this line. Otherwise,

set to Inactive.

 Forward reason:

 Unconditional forward: forward the call all the time  Busy forward: Forward the call when phone is busy.  No answer forward: forward the call when the call does not

answered after no answer timeout.

 Forward TEL: The forward telephone number for the selected

reason

 No answer timeout (seconds): The no answer timeout will be used

(default is 120 sec)

 Call waiting: Enable call waiting or not.

features, the second incoming call will be rejected.

 Reject Anonymous Call: Reject the anonymous incoming call or not  Hot line: Enable to disable hot line feature  Hot line TEL: The number to be dialed automatically after the user

pickup the phone.

 Polarity Reversal generation: Enable Polarity Reversal for FXS as

billing signal or not. When a FXS calls to VOIP and answered by the VOIP, 25xx will generate reverse signal to FXS as a billing start. When VOIP side disconnect first, 25xx will reverse back as a billing stop

signal

 Current Drop generation: Enable current drop (0 voltage) when

VOIP is disconnected or not.

 Input(Encode)Gain: Adjust the volume from FXS/FXO to VOIP

(default is 0 db)

 Output(Decode)Gain: Adjust the volume from VOIP to FXS/FXO

(default is 0 db)

 FAX Relay: Enable T.38 Fax Relay or T.30 Fax Bypass or not.  Voice mail subscription: enable voice mail subscription (MWI) or

not.

 Caller ID mode:

 Inhibit: don’t send caller ID to analog phone.  Transparent: send caller ID to analog phone.

 SIP caller ID mode:

 Inhibit: don’t send caller ID to VOIP SIP  Transparent: send caller ID to VOIP SIP

 Register Type:

 Register: register to proxy. If it is not registered to SIP proxy,  Predefine: When it is set to predefine, 25xx will not send  Internal: When it is set to internal, 25xx does not send

(T.30 Fax Bypass only supports G711a law)

the FXS line still can use SIP trunk for VOIP call.

When disable call waiting

 TEL No: The registrar telephone number  User ID: The SIP user ID for register and call making  User Password: The SIP password for register and call making  Display Name: The SIP display name

SIP Proxy>

The SIP proxy server defined here is dedicated used for FXS lines.

 Domain: The SIP domain for register or call making  Primary proxy server: Primary SIP registrar server address  Primary proxy server port: Primary SIP registrar server port

number

 Outbound Proxy server: Primary outbound proxy server address  Outbound Proxy server port: Primary outbound proxy server port

number

 Primary Proxy server keep Alive: using through NAT and keep the

port.

 Keep Alive Time(sec): Specify of times send sip register message to

proxy server.

 Secondary Proxy: Enable secondary proxy or not. When enable it,

the primary and secondary proxy will be registered at the same time.

 Secondary proxy server: Secondary SIP registrar server address  Secondary proxy port: Secondary SIP registrar server port number  Secondary outbound Proxy server: Secondary outbound proxy

server address Secondary

 Outbound Proxy server port: Secondary outbound proxy server

port number

 Register Expire: SIP register time to live  Secondary Proxy server keep Alive: using through NAT and keep

the port.

 Keep Alive Time(sec): Specify of times send sip register message to

proxy server.

Caller ID>

The call ID stand for the phone

 Caller ID Mode: Caller ID mode to be used for phone (FSK

Bellcore/FSK ETSI/DTMF)

 Polarity Reverse before caller ID: start polarity reverse before

send the caller ID

 Dual tone before caller ID: Send dual tone before caller ID (for FSK

ETSI use only)

 Caller ID present: The timing to send the caller ID

(Before first ring/after first ring/after first short ring)

 DTMF caller ID start digit: specify the DTMF caller ID start digit

(default is D, the range is A to D)

 DTMF caller ID stop digit: specify the DTMF caller ID start digit

(default is C, the range is A to D)

Others>

Flash time and current drop generation/detection time

 Min flash time(80~800msec):

If the phone-set’s flash time is shorter than the Flash Low setting, the flash will be ignored.

 MAX flash time (80~800msec): Specify the value of the flash

(high), setting, the flash will be handled as hang-up.

 Current Drop Times (msec): Specify the value of the current drop

times (generate – for FXS / detect – for FXO).

If the phone-set’s flash time is longer than the Flash high

Specify the value of the flash (low),

SIP Trunk:

The administrator needs to set the SIP trunk for VOIP outgoing call and incoming call. There are up to 4 SIP trunk can be used for whole system.

 Trunk ID: SIP trunk ID 1 to 4  Register Type: Register type is predefine or register  TEL No: The Tel no for the SIP account  Proxy Server: The SIP proxy server  Proxy Server port: The SIP proxy server port  Outbound Proxy: The SIP outbound proxy sever  Outbound Server Port: The SIP outbound proxy server port

Create SIP Trunk>

>For FXS

>For FXO

 Trunk ID: SIP trunk ID 1-4  Register Type: Whether this account need register or not

 Register: When it is set to register, 25xx will send

 Predefine: When it is set to predefine, 25xx will NOT send

 Domain: The SIP domain for register or call making  Proxy Server: SIP registrar server address  Proxy Server Port: SIP registrar server port number  Outbound Proxy Server: outbound proxy server address  Outbound Proxy server port: outbound proxy server port number  Register Expires: the default register expired for negotiation  TEL No: The registrar telephone number  User ID: The SIP user ID for register and call making  User Password: The SIP password for register and call making  Display Name: The SIP display name  Reject Anonymous Call: Reject the anonymous call  Outgoing Caller ID: The outgoing SIP caller ID mode.

-Display Name: The display name will be set according to the following type.

None: No display name will be used PSTN caller ID: The display name will be the collected PSTN

caller ID

SIP display name: The display name will be the Display

Name set in this SIP trunk.

FXO Tel NO: The display name will be the incoming FXO’s TEL

No set on FXO lines.

-User ID: The SIP caller ID will be used according to the following type.

 SIP user ID: If the SIP user ID is set, the SIP user ID set in

this SIP trunk will be used and the domain/SIP proxy will be the host part. The SIP FROM header’s URL will be the SIP_User_ID@Domain or SIP_User_ID@SIP_Proxy_Server.

 PSTN caller ID: If the PSTN caller ID will be used in SIP URL,

the SIP FROM header’s URL will be PSTN_Caller_ID@local_IP_address/

 FXO Tel NO: If the FXO Tel NO will be used in SIP URL, the

SIP FROM header’s URL will be FXO_Tel_NO@local_IP_address.

The following guideline could be used for most of case:

1. If the 25xx in SIP proxy was handled as a gateway,

please set both the display name and User Id to be “PSTN caller ID”.

2. If the Dynamix 25xx in SIP proxy was handled as

a subscriber, please set the display name to “PSTN caller ID” and user ID to “SIP User ID”.

 For DNIS is Register TEL: Using FXO port that have an incoming call

from VOIP dial to PSTN, it can be chosen 2 methods. ( FXO only use )

 1 stage dialing: If match a sip trunk number or not, it will

direct to dial out that number before Making DM and route plan.

 2 stage dialing: If match a sip trunk number, it will auto

answer the call and playing dial tone.

 Keep Alive: Enable or Disable it.  Keep Alive Time (sec): Specify interval time to send sip register

message to proxy server.

Route Plan:

The core of

Dynamix

25xx is the routing policy. The policy is based on incoming call type/target, length and prefix to determinate the outgoing call process. For VOIP incoming call, it can send to FXO or FXS interface and vice versa.

For FXO interface>

For FXO interface, it could be routed to VOIP and vice versa.

 Incoming Call Type: Incoming call type (FXO or VOIP)  Matched Prefix: matched DNIS (called number) prefix  Matched Incoming List: matched DNIS incoming interface target  Matched Length: matched DNIS (called number) length  Outgoing Type: The outgoing call type (FXO or VOIP)

For FXS Interface>

For FXS interface, it could be routed to VOIP and vice versa. You can ignore the routing plan if you don’t need it for FXS interface.

 Incoming Call Type: Incoming call type (VOIP or FXS)  Matched Prefix: matched DNIS (called number) prefix

 Matched Incoming List: matched DNIS incoming interface target  Matched Length: matched DNIS (called number) length  Outgoing Type: The outgoing call type (FXS or VOIP)

Create Route Plan>

Click Route Plan and Click new to create a new routing policy.

For FXO>

 Incoming Call Type: Incoming call type

 VOIP: The incoming SIP call type

 FXO: The PSTN incoming call type  Matched Prefix: matched DNIS (called number) prefix  Matched Incoming List: matched DNIS incoming interface target

 For VOIP incoming call type, the incoming target will be

the SIP trunk ID. Only the call from the selected SIP Trunk will be accepted for this route.

 For PSTN incoming call type, the incoming target will be

the line ID (L1 to L4). Only the call is coming from the selected line will be accepted for this route.

 Matched Length: matched DNIS (called number) length. For ignoring

the length, please set to 0.

 No Answer Timeout: How long the hunting will continue to next

when the called target doesn’t answer.

Create Route Plan>Primary Route

 Outgoing Type: Outgoing call type (FXO or VOIP)

 Hunting Type: The hunting method will be used for this route.

 Priority Ring: The call will be hunted based on the

routing list order one by one.

 Cyclic Ring: The call will be hunted based on the cyclic

basis. This is the recommended method.

 Routing List:

The routing target list will be used for this route.

 DM Group: Select DM group 1 to 4 in case it requires a DM (for

example remove the prefix) before to make the call.

Create Route Plan>Backup Route

 Backup Route Active: Active the backup route or not.  Outgoing Type: The backup route outgoing call type.  Hunting Type: The hunting method will be used for this route. Please

refer to the Primary Route.

 Routing List: The backup routing target list will be used for this route.  Route DM Group: Select DM group 1 to 4 in case the backup required

the DM before to make the call. The DNIS is unchanged by the primary route DM and same as the DNIS before routing. For example, the DNIS is 886282265699 and primary DM group remove 886 and use it (DNIS = 282265699) to make call. When backup route is started, the DNIS is still unchanged as 886282265699. This makes the DM easy to predict and implement.

2 special default route, “VOIP Default Route” and “FXO default Route”, are used as the default routing when there is no any other routing are matched. It is not recommended to disable these 2 default route. The FXO default route is used when a FXO incoming call’s default routing. VOIP default route is used for a VOIP incoming call’s default routing.

For FXS>

 Incoming Call Type: Incoming call type

 VOIP: The incoming SIP call type

 FXS: The FXS extensions incoming call type  Matched Prefix: matched DNIS (called number) prefix  Matched Incoming List: matched DNIS incoming interface target

 For VOIP incoming call type, the incoming target will be

the SIP trunk ID. Only the call from the selected SIP Trunk will be accepted for this route.

 For FXS incoming call type, the incoming target will be

the line ID (T1 to T4). Only the call is coming from the selected line will be accepted for this route.

 Matched Length: matched DNIS (called number) length. For ignoring

the length, please set to 0.

 No Answer Timeout: How long the hunting will continue to next

when the called target doesn’t answer.

Create Route Plan>Primary Route

 Outgoing Type: Outgoing call type (FXS or VOIP)  Hunting Type: The hunting method will be used for this route.

 Priority Ring: The call will be hunted based on the

routing list order one by one.

 Cyclic Ring: The call will be hunted based on the cyclic

basis. This is the recommended method.

 Routing List:

The routing target list will be used for this route.

 DM Group: Select DM group 1 to 4 in case it requires a DM (for

example remove the prefix) before to make the call.

Create Route Plan>Backup Route

 Backup Route Active: Active the backup route or not.  Outgoing Type: The backup route outgoing call type.  Hunting Type: The hunting method will be used for this route. Please

refer to the Primary Route.

 Routing List: The backup routing target list will be used for this route.  Route DM Group: Select DM group 1 to 4 in case the backup required

2 special default route, “VOIP Default Route” and “FXS default Route”, are used as the default routing when there is no any other routing are matched. It is not recommended to disable these 2 default route. The FXS default route is used when a FXS outgoing call’s default routing. VOIP default route is used for a VOIP incoming call’s default routing.

States:

25xx provides the system status here.

Device States>

 Model: The model number  MAC-Address: The MAC address of 25xx  Network Type: The Network Interface Type Settings  IP-Address: IP address is using  Firmware: The firmware version and release information

Line States>

This page shows the each line’s current status.

For FXO>

For FXS>

 Line: L1 to L4  Account: The line account.  Registered: The line status for registered or not.  Call State: The line status for this line  Refresh Interval (second): The time to refresh the status

SIP Trunk States>

 Account: SIP trunk account  Registered: The SIP trunk register status  Concurrent Call: The concurrent calls are used for this SIP trunk  Refresh Interval (second): The time to refresh the status

Maintenance:

25xx can be managed by this management page for upgrading firmware or reset.

 Backup: Backup the system settings for restoring purpose  Restore: Restoring the backup setting back to 25xx  Reset to Default: Reset system setting to factory default  Quick-Reset: Warm Reset without reboot 25xx  Reboot: reboot 25xx

Firmware Update>

This maintenance page provides the firmware upgrade features.

 Firmware Update: Upgrade the new firmware through web page

Appendix A --- Call Processing Flow (

Dynamix FXO-04

)

Appendix B --- System Recovery

25xx use dual firmware image to ensure the system stabilities. In most of case, you will not encounter the system failed to boot issue. Normally, the user should be able to use Web page to login and upgrade the firmware through it. If you are not able to do it, please follow the following steps for recovery.

1. Start the 25xx and to check the STATUS led is up or not. If STATUS led is ON,

please press the reset button for 5 seconds to reset to default. After all LED are light up, the system is back to factory settings.

2. Change your PC IP address to

mode.

3. Connect your PC to WAN port and use http://192.168.123.123 to upgrade the

firmware. Make sure you are using Micrsoft IE 6 or later version. DO NOT support FireFox Web browser.

4. If you cannot login to the web page through

line windows and type “telnet 192.168.123.123”. If you can see the following display, go to the next step. Otherwise, please contact Welltech FAE for RMA (Repair).

5. Prepare a TFTP server for firmware download as follows

- download tftp server

http://tftpd32.jounin.net/tftpd32_download.html

- start tftp server

192.168.123.111 and network set to fix IP address

192.168.123.123. Open a command

- download the firmware into tftp data directory

6. In the telnet terminal, do the following command

- 1. __dmctw

- 2. cd /config_fs

- 3. rm -f wg25*.bin

- 4. tftp –g –r wg25.2.0.bin 192.168.123.111

- 4. copy firmware successfully

- 5. reboot

7. Check whether the system was recovered or not

Appendix C --- HTTP auto provisioning

Get the http provision packet from Welltech and start the provision as follows:

Step 1: build mac list for mass configuration file generation

Please open the “wg2504 MAC.csv or wg2540 MAC.csv” which was provided from Welltech by using Microsoft Excel. You can refer to the picture below. Normally, you should get all required configuration mac list from Welltech and use it for configuration file generation if you have ordered a bulk of quantity.

For FXS>

The wg2504 MAC.csv contains most frequently changed parameters as follows:

MACAddress: Dynamix 25xx MAC Address fxs1.displayname ~ fxs4.displayname: display name for each line fxs1.password ~ fxs4.password: user password for register to SIP proxy

for each line

fxs1.telno ~ fxs4.telno: telephone number for each line fxs1.userid ~ fxs4.userid: user id for register to SIP proxy for each line

For FXO>

The wg2540 MAC.csv contains most frequently changed parameters as follows:

MACAddress: Dynamix 2540 MAC Address Siptk1.displayname ~ siptk4.displayname: display name for each line Siptk1.userid ~ siptk4.userid: user id for register to SIP proxy for each line siptk1.password ~ siptk4.password: user password for register to SIP

proxy for each line Siptk1.telno ~ siptk4.telno: telephone number for each line

Please save and close it.

Step 2: create a template configuration file

Open the “wg2504 Parameter.txt” or “wg2540 Parameter.txt” which was provided by Welltech and make the required changes. Please make some changes for those provision and SIP proxy settings at least. For details, please refer to the comments of “wg2504 Parameter.txt” or “wg2504 Parameter.txt”.

Step 3: Make the changes for “wegencfg.ini” as follows if it is necessary

# Template File BaseFile=.\wg2504 Parameter.txt # MAC list file

ListFile=.\wg2504 MAC.csv # 0: Off, 1: On Encrypt=0

# Template File

BaseFile=.\wg2540 Parameter.txt

# MAC list file

ListFile=.\wg2540 MAC.csv # 0: Off, 1: On Encrypt=0

Step 4: Generate the individual configuration file.

Double click the “wtgencfg.exe”, it will generate the configuration file for each MAC list in “MAC address.cfg” as the following pictures.

Step 5:

Put the “*.cfg” file into http or ftp directory. Set the provision settings in

Dynamix 25xx and reboot to test it. You can use the hfs for http file server. It

can be downloaded from this WebPage http://www.rejetto.com/hfs/.

Dynamix FXS-04, FXO-04, 2504, 2540 User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual