Grandstream Networks HT503 user manual VoIP-to-PSTN Calls, PSTN-to-VoIP Calls

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To receive PSTN calls, pick up the phone when it rings;

To complete a PSTN call, press the PSTN access code (*00 is default, or any number configured in the web configuration) to switch to the PSTN line, listen for a dial tone, then dial the number.

If the 503 loses power or lost registration with SIP server, device will switch to mode when PSTN line will be transparently connected directly to phone connected to FXS port. It will function as a jack, enabling a direct connection to the PSTN Line.

VoIP-to-PSTN Calls

This function is available using the FXO port. The FXO port functions as a bridge between the Internet and PSTN. The user can remotely use a PSTN line to initiate a call.

TO MAKE A VOIP-TO-PSTN CALL:

1.Dial the FXO SIP account phone number to establish the VoIP session. The caller will hear the ring back tone once. Then the caller hears either a special continuous tone or a dial tone. The special continuous tone is played if the pin code is configured, otherwise, the caller will hear a dial tone.

2.Enter the PIN code (if configured under the BASIC configuration page). The caller will hear a dial tone and be connected to the PSTN line if the PIN code is valid. If the PIN code is invalid, the continuous tone is played to prompt caller to enter the PIN code again. The user may try up to 3 times to enter a correct PIN code. After three (3) tries, the HT503 hangs up.

3.After the caller hears a dial tone from PSTN line, the caller can place the next call.

4.The user can hit the # key to identify the end of the pin code or wait 4 seconds for a new dial tone and then dialing the PSTN number.

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 PSTN” 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

PSTN-to-VoIP Calls

This function is available using the FXO port. The FXO port functions as a bridge between the Internet and PSTN and enables calls to be passed from the PSTN network to VoIP. The user can make VoIP calls remotely by dialing into the FXO line port on HT503.

To Make a PSTN-to-VoIP Call:

1.Make an incoming call to the PSTN line on FXO port. The phone will ring for 4 times by default (this setting is configurable on the FXO port configuration page).

Grandstream Networks, Inc.

HT503 User Manual

Page 13 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 Equipment Packaging Connecting the HT503WAN LED Power LEDLAN LED PHONE/ Line LEDSoftware Features Overview LED Hardware SpecificationUnderstanding HT503 Voice Prompt To reset the HT to take affect the new IP addressMain Menu Phone or Extension Numbers Placing a Phone CallCall Waiting Call HoldCall Transfer Pstn Pass Through Way ConferencingPSTN-to-VoIP Calls VoIP-to-PSTN CallsForward Calls to Pstn Route Calls to PstnOne Stage Dialing Forward Calls to VoIPFax Support Blind Transfer Enable Srtp Disable SrtpFlash/Hook Static IP Mode Configuring HT503 through Voice PromptAccess the Web Configuration Menu Configuring HT503 with Web BrowserDND NATFXS FXOMTZ+6MDT+5 DMZ IP Firmware Upgrade Admin PasswordLayer 3 QoS Layer 2 QoSACS URL HTTP/HTTPSLife Line Mode Disable Direct IPConfiguration Lock KeypadAuthentication Password Authenticate IDDNS mode Unregister on RebootDisable Dtmf SIP T1 TimeoutEnable Call Features Disable Call WaitingDial Plan Prefix Special FeatureUse # as Dial key Dial Plan Dial Plan RulesVAD Slic Setting Srtp ModeDisable Line Echo Canceller LECOutgoing Call Without Authenticate PasswordSIP registration failure Retry wait timeDian Plan Negotiation Proxy RequireInvite Ring-No-Answer Timeout Preferred VocoderFSK Caller ID minimum Caller ID SchemeRX Level dB FSK Caller ID SeizureEnable Pstn Disconnect Enable CurrentPstn Ring Thru Delay First Digit Timeout secRebooting from Remote Saving the Configuration ChangesConfiguration through a Central Server Software Upgrade Firmware Upgrade through TFTP/HTTP/HTTPSFirmware and Configuration File Prefix and Postfix Configuration File DownloadManaging Firmware and Configuration File Download Restore Factory Default Setting
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