| incoming INVITE |
| call will be rejected. If this option is enabled, the device will not be able to make direct |
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| IP calls. |
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| SIP T1 Timeout |
| T1 is an estimate of the |
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| If the network latency is high, select larger value for reliable usage. |
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| SIP T2 Interval |
| Maximum retransmission interval for |
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| DTMF Payload Type |
| Sends DTMF using RFC2833 |
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| Preferred DTMF method |
| The HT503 supports up to 3 different DTMF methods including |
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| (in listed order) |
| (RFC2833) and via Sip Info. User can configure DTMF method in a priority list. |
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| Disable DTMF |
| Default is No. If set to yes, use above DTMF order without negotiation |
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| Negotiation |
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| Proxy Require |
| SIP Extension to notify SIP server that the unit is behind a NAT/Firewall. |
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| Use NAT IP |
| NAT IP address used in SIP/SDP message. Default is blank. |
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| Ring Timeout |
| Sets the time in which an incoming from PSTN call will stop ringing when not picked up. |
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| Early Dial |
| Default is No. Use only if proxy supports 484 response. This parameter controls |
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| whether the phone will send an early INVITE each time a key is pressed when a user |
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| dials a number. If set to “Yes”, an INVITE is sent using the |
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| far. Otherwise, no INVITE is sent until the |
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| seconds have elapsed. The “Yes” option should be used ONLY if there is a SIP proxy |
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| configured and the proxy server supports 484 Incomplete Address response. |
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| Otherwise, the call will likely be rejected by the proxy (with a 404 Not Found error). |
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| Note: This feature is NOT designed to work with and should NOT be enabled for direct |
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| Dial Plan Prefix |
| Sets the prefix added to each dialed number. |
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| Use # as Dial Key |
| This allows users to configure the # key as the “Send” (or “Dial”) key. If set to “Yes”, “#” |
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| will send the number. In this case, this key is essentially equivalent to the “Dial” key. If |
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| set to “No”, the “#” key can be included as part of a number. |
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| Dian Plan |
| Dial plans work only for incoming calls from PSTN network. In case unconditional call |
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| forward to VoIP is configured, dial plan feature will not work. In case of normal dialing |
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| to VoIP, after dialing PSTN number, |
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| If using the ‘hop on/hop off’ feature, the dial plan rules affect only the last called number |
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| (i.e. the number called after receiving dial tone from the ATA). |
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| Dial Plan Rules: |
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| 4. Accept Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d |
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| 5. Grammar: x - any digit from |
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| a. xx+ - at least 2 digits number; |
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| b. xx. – at least 2 digits number; |
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| c. | ^ - exclude; |
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| d. |
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| e. [147] - any digit 1, 4, or 7; |
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| f. | <2=011> - replace digit 2 with 011 when dialing |
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| • Example 1: {[369]11 1617xxxxxxx} – |
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| Allow 311, 611, 911, and any 10 digit numbers of leading digits 1617 |
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| • Example 2: {^1900x+ <=1617>xxxxxxx} – |
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| Block any number of leading digits 1900 and add prefix 1617 for any dialed 7 |
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| digit numbers |
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| • Example 3: |
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| Allow any length of number with leading digit 2 and 10 |
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| leading digit 1 and leading exchange number between 2 and 9; If leading digit |
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| is 2, replace leading digit 2 with 011 before dialing. |
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| 6. Default: Outgoing - {x+} |
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| Example of a simple dial plan used in a Home/Office in the US: |
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| Grandstream Networks, Inc. |
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| HT503 User Manual | Page 31 of 38 |
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| Firmware 1.0.4.2 | Last Updated: 06/2011 |
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