Telephony Settings. D-Link DVG-2016S

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Choose a DDNS Server: The current system allows users to choose either DynDNS、TZO、3322.org、

PeanutHull or a private server. Please apply for a user account before choosing a service provider.

Server address: Sets up the IP address or URL (Uniform Resource Locator) of the DDNS

Server.

Hostname: The URL of the system (or NAT) – apply from a domain name registration providers.

Login ID and Password: The ID and password are used to log into the DDNS server.

Behind NAT: Select only when the system is set up under NAT.

Note: If the Gateway is set up under NAT, then enter the hostname into the NAT IP/Domain that is the same with Hostname of DDNS.

Example:

NAT

DDNS

Telephony Settings

FXS Caller ID Generation: Select this option to enable the caller ID display function. When enabled, the caller’s phone number will be displayed on your phone set when the call comes through. FSK is preferred in some countries.

FSK Caller ID Type: Select FSK type. In most cases, Bellcore is preferred in North America and ETSI in Europe.

Anonymous Caller ID (CLIR): When enabled, the caller’s phone set will not display your number.

Note: If you register the gateway to a Proxy, you may be unable to make a call. This is due to the fact that the gateway doesn’t send the number for authorization.

Enable: Enable a line; if some lines are not used, disable them (Pause Function) to avoid unnecessary waiting when an incoming call is diverting to this line.

Hotline Functions

FXS port: When the user picks up the phone, the gateway automatically dials your assigned hotline number. When in hotline mode, other lines cannot be used.

Hot Line No.: Enter the hot line number for an automatic dialing function.

Warm Line: When the warm line function is in use, user can dial a number. Otherwise the system will divert incoming calls from an outside line to the Hot Line Number after a set wait time.

FXS Group: Select group hunting when there is an incoming call, the gateway will automatically assign an unassigned call according to Hunting Priority. If Line 2 does not want to be set as an assigned line to receive any inbound calls, the function can be disabled.

Users can also use the Up or Down key to adjust hunting priority.

Enable FAX: Enable this line to detect if there is a FAX tone to transfer the Codec.

Ring Time Limit(10 - 600secs)：The timeout to cancel a call when no one answers.

Enable End of Digit Tone：The gateway will play a “Beep-Beep” tone to notify the call is in progress.

VoIP Calling Notification: The gateway will play a tone to notify the call is through VoIP.

Force Calling Thru PSTN code: Dial the code to get a PSTN line for dial out. For example: If you would like dial “23456789” through PSTN and Force Calling Thru PSTN code is *33, just dial “*33 23456789”

Early Media Treatment: If it is disabled, the system will send RTP immediately when the connection with Proxy is set up. The default is enabled. If communicating with other

Gateway has problem, please disable this function.

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SIP

Hunting/Ring: It is able to set FXS group hunting using simultaneous ring or sequential ring.

Sequential Ring Time: To set the ring time of each port, when sequential ring is chosen.

All Call through OutBound Proxy：An outbound proxy server handles SIP call signaling as a standard SIP proxy server would. Furthermore, it receives and transmits phone conversation traffic (media) in between two talking gateways. This option tells the gateway to send and receive all SIP packets to the destined outbound proxy server rather than the remote gateway. This helps VoIP calls to pass through any NAT protected network without additional settings or techniques. Please make sure your VoIP service provider supports outbound proxy services before enable it.

Session Expiration: It is to avoid the billing of abnormal dropping the call because of

Internet. The default is disabled.

Session Refresh Request: to send the packet of UPDATE or re-INVITE to

Session Refresher: It is the gateway’s role in Session Timer. UAS is an originator, and UAC is a replier.

Enable P-Assert: It is for caller id protect.

Privacy Type: Privacy requested for Third-Party Asserted.

E.164

SIP Message Resend Timer Base: SIP packet will resend if response didn't arrive in the base time set in this column. It will send again at "base time" * 2, and send again at "base time" *2 *2. The max of resend time is 4 sec. Resend will stop/restart when total resend

20sec has reached.

Max. Response Time for Invite: If the destination does not reply in the set time, this call is failed.

Invite URL need ‘user=phone’: There is ‘user=phone’ in invite packet.

Reliability of Provisional Responses: Provide information on the progress of the request processing if ticked.

Compact Form: It decreases the size of SIP header if ticked.

International Call Prefix Digit: Enter the International call prefix.

Country Code: Users please select the desired country code.

Long Distance Call prefix Digit: The long-distance prefix digit for making a long-distance call.

Area Code: Please enter the area code.

E.164 Numbering: To invite Proxy to follow the E.164 rule. It depends on the Proxy. If you fail to make a call, please contact your ITSP .

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Enable Support of SIP Proxy Server / Soft Switch: Enable the functions to inter-work with

Proxy Server / Soft Switch. When SIP Proxy 1 and 2 are enabled, the system will register to

SIP Proxy 2 after all lines are failed to register to SIP Proxy 1. SIP Proxy 2 is a backup system.

Proxy Server IP/Domain: Enter the Proxy Server IP address or URL (Uniform Resource

Locator). You can set 3 redundant Proxy spread by semicolon.

EX: 61.123.231.1;12.34.56.78;proxy.sip.sip

Proxy Server Port: Enter the Proxy Server listen port number. (The factory default value is

5060)

Proxy Server Realm: Enter the correct registered Proxy Server Realm name to avoid registration failure. If you fail to make a call, please contact your ITSP.

TTL: Enter the desired time interval at which the gateway will report to you Proxy Server.

SIP Domain/Use Domain to Register: Enter the correct SIP domain to avoid registration failure (it is not necessary to set with some Proxy Servers). If you enable “Uses Domain to

Register” the gateway will register to Proxy with the domain name you filed. Else, the gateway will register to a Proxy with the IP it resolves. If you fail to make a call, please contact your ITSP .

Bind Proxy Interval for NAT: This function is able to keep the binding is existed when the gateway is behind NAT and SIP Proxy is not able to keep the binding.

Initial Unregister: After rebooting, it is unregistered first and then do the general registry process.

Enable Message Waiting Indication: The system will play a tone to remind users that there are messages in SIP Server.

Proxy-Require: Some SIP Sever need SIP UA to add this header to it's sip message.

FXS Representative number registers to Proxy:

Assuming that your registered ID and password are individual, the settings should be as above.

FXS Representative Number: Register all FXS ports as a hunting group.

Register: Register to Proxy if ticked.

Invite with ID / Account: DVG-2016S can be invited to a VoIP trunk gateway w/o register to a Proxy. Please contact your ITSP

NOTE: Please ensure that if Proxy Server allows one account for many ports using before using representative number to register.

Each line registers to Proxy independently:

Invite with ID / Account: The gateway can be invited to a VoIP trunk gateway w/o register to a Proxy. Please contact your ITSP

As there are various Proxy Server providers, our company has designed the gateway to be compatible with them, and according to RFC standards. If any registration problem occurs, please consult your Proxy

Server provider.