| Enable Current |
| Default is Yes. This value should be used in case the PSTN provider uses line power |
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| Disconnect |
| drop to indicate call completion to the end point. In this case the HT503 will search for |
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| a power drop for a preconfigured time frame to disconnect such calls from a VoIP |
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| extension. |
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| Current Disconnect |
| This is a preconfigured value of duration for a line power drop used by specific service |
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| Threshold (ms) |
| providers. For example, for a configured value of 500ms the device will ignore any |
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| random voltage drops on the line if duration of such drop is less than 500ms and the |
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| call will NOT be considered as terminated. This is useful to prevent unnecessary call |
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| drops in some low quality PSTN lines. |
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| Enable PSTN Disconnect |
| If set to Yes, arrived Busy Tone is used as the disconnect signal. |
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| Tone Detection |
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| PSTN Disconnect Tone |
| In certain countries, the central office will send a special busy tone to indicate when a |
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| call is disconnected from the remote side. User can |
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| ATA. The user should know the frequency values and cadences of these tones. |
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| Here is an example for the syntax for a busy tone in the U.S.A: |
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| (Syntax: f1=freq@vol, f2=freq@vol, |
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| (Note: freq: 0 - 4000Hz; vol: |
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| (Default: Busy Tone - |
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| AC Termination Model |
| You can select the AC termination by Country or by Impedance. |
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| 15 Countries are selectable in this version of the F/W. |
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| Select the Impedance used by the PSTN service provider. |
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| Number of Rings |
| Default is 4. This setting specifies number of phone rings (on the phone connected to |
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| the FXS port) before a PSTN incoming call is bridged to VoIP |
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| Note: The number of rings feature serves as a PSTN answer delay, and should be set |
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| to a larger value to allow enough time for the HT503 to decode the Caller ID signal set |
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| by the central office. |
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| PSTN Ring Thru FXS |
| If Yes, the phone connected to the FXS port will ring a configured amount of times (see |
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| above). If not, the phone connected to the FXS port will not ring. |
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| PSTN Ring Thru Delay |
| If the PSTN Ring Thru Delay is set to Yes, all incoming PSTN calls through FXO will |
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| (sec) |
| ring the phone connected to the FXS port, after this delay or after caller id is detected |
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| (whichever comes first). |
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| DTMF Digit Length (ms) |
| Digit length and Dial Pause are port digit dialing configurations; FXO needs to dial out |
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| digits for VOIP to PSTN 1 stage calls, and unconditional call forward to PSTN, and |
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| route to PSTN. Digit Length is the play time for each digit. |
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| Note: In order to receive the caller ID information, the delay should be set to a value |
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| larger than the delay required to complete the PSTN caller ID delivery. |
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| DTMF Dial Pause (ms) |
| Dial pause is the time between 2 digits for the same scenario as explained above. |
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| First Digit Timeout (sec) |
| Used for PSTN to VoIP calls. PSTN users need to enter the FIRST digit within the first |
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| digit timeout period. Otherwise the call will be dropped. |
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| Inter Digit Timeout |
| When dialing from the PSTN to VoIP, subsequent digits have to be input within the |
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| period of |
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| Wait for Dial Tone |
| Wait for Dial tone is used for one stage VoIP to PSTN calls. If set to Yes, the device |
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| will first obtain a PSTN line and a dial tone from a central office. After obtaining the dial |
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| tone, the digits dialed will be sent to the central office. |
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| Stage Method (1/2) |
| This configuration is applicable for VoIP to PSTN calls and indicates one or two stage |
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| dialing methods. |
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Grandstream Networks, Inc. | HT503 User Manual | Page 34 of 38 |
| Firmware 1.0.4.2 | Last Updated: 06/2011 |
