Grandstream Networks HT503 user manual Route Calls to Pstn, Forward Calls to Pstn

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2.If no one answers the call after 4 rings (default configuration), then the caller hears either a special continuous tone (prompting a PIN number) or a dial tone.

3.Enter a valid PIN (if configured under the BASIC configuration page). The caller will hear dial tone and be bridged to VoIP. If an incorrect PIN is input, the continuous tone prompts caller to enter a valid PIN. The caller may try 3 times to enter a valid PIN, if it is invalid the HT503 will hang up.

4.The caller can dial a VoIP number followed by # (or wait for 4 seconds); the VoIP call will be initiated from the SIP account configured on the FXO port.

NOTE:

Users can choose whether or not to apply password protection for VoIP-to-PSTN calls. A PIN (Pin for PSTN calls) consists of up to 8 numeric digits and can be configured using the BASIC SETTINGS of the web configuration page. By default, there is no password protection. (I.e. there is no authentication required for callers on the use of PSTN line through HT503).

When a PIN is configured for VOIP-to-PSTN call flow, the VoIP device that calls into the HT503 FXO account needs to configure RFC2833 or SIP Info for DTMF digit transmission.

The special continuous tone is the prompt to enter a valid PIN code. If a caller doesn’t enter a valid PIN, the HT503 times out after 10 seconds. Users may press the “#” key to indicate the end of an input or wait 4 seconds.

On the web configuration page, if the “Forward to VoIP” is configured, the second stage dialing format is eliminated, so after dialing into the FXO SIP account number, the PSTN number will be called automatically

Route Calls to PSTN

The FXO port enables access to the PSTN network. By default, the HT503 is in VoIP mode at off-hook. If “Route Call to PSTN” is configured, certain calls will be initiated from the FXO PSTN line port. This call feature is especially useful for emergency calls or local telephone calls.

To use this feature, users need to specify a special rule using the dial plan parameter located under FXS Port configuration page. If the dialed digits match the specified prefix, outbound calls will be initiated from the PSTN line.

Note: The route to PSTN feature is only applicable to a phone connected to the FXS Port. The configuration is done using the “dial plan” feature under the FXS tab. An example of the configuration is {L: 911x+}. This shows that only calls that start with 911 are immediately forwarded to the PSTN line. All other numbers will not be routed to the PSTN. An normal # would be: {L: 617x+x+} or {x+ L: 617x+}

For example, if “Route Call to PSTN” is configured as {L: 626x+}, all outgoing calls starting with 626 will be initiated from the PSTN line.

Forward Calls to PSTN

Any VOIP call may be forwarded to a specified PSTN number. FXO port should be registered with some preconfigured number (for example 1111). Any VoIP extension can dial this FXO account number and will be automatically forwarded to preconfigured PSTN extension.

For example, if the end-user has configured a cell phone number in the field “Forward to PSTN” under BASIC SETTINGS configuration page, all calls will be forwarded to the cell phone number after 4 rings.

Grandstream Networks, Inc.

HT503 User Manual

Page 14 of 38

 

Firmware 1.0.4.2

Last Updated: 06/2011

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Contents Grandstream Networks, Inc Safety Compliances Warranty Table of Figures Welcome Connecting the HT503 Equipment PackagingLAN LED Power LEDWAN LED PHONE/ Line LEDSoftware Features Overview Hardware Specification LEDMain Menu To reset the HT to take affect the new IP addressUnderstanding HT503 Voice Prompt Placing a Phone Call Phone or Extension NumbersCall Transfer Call HoldCall Waiting Way Conferencing Pstn Pass ThroughVoIP-to-PSTN Calls PSTN-to-VoIP CallsRoute Calls to Pstn Forward Calls to PstnFax Support Forward Calls to VoIPOne Stage Dialing Flash/Hook Enable Srtp Disable SrtpBlind Transfer Configuring HT503 through Voice Prompt Static IP ModeConfiguring HT503 with Web Browser Access the Web Configuration MenuFXS NATDND FXOMTZ+6MDT+5 DMZ IP Layer 3 QoS Admin PasswordFirmware Upgrade Layer 2 QoSHTTP/HTTPS ACS URLConfiguration Disable Direct IPLife Line Mode Lock KeypadDNS mode Authenticate IDAuthentication Password Unregister on RebootEnable Call Features SIP T1 TimeoutDisable Dtmf Disable Call WaitingUse # as Dial key Special FeatureDial Plan Prefix Dial Plan Dial Plan RulesVAD Disable Line Echo Srtp ModeSlic Setting Canceller LECSIP registration failure Authenticate PasswordOutgoing Call Without Retry wait timeNegotiation Proxy Require Dian PlanTimeout Preferred Vocoder Invite Ring-No-AnswerRX Level dB Caller ID SchemeFSK Caller ID minimum FSK Caller ID SeizurePstn Ring Thru Delay Enable CurrentEnable Pstn Disconnect First Digit Timeout secConfiguration through a Central Server Saving the Configuration ChangesRebooting from Remote Firmware Upgrade through TFTP/HTTP/HTTPS Software UpgradeManaging Firmware and Configuration File Download Configuration File DownloadFirmware and Configuration File Prefix and Postfix Restore Factory Default Setting
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