Grandstream Networks HT503 Incoming Invite, SIP T1 Timeout, SIP T2 Interval, Dtmf Payload Type

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incoming INVITE

 

 

call will be rejected. If this option is enabled, the device will not be able to make direct

 

 

 

 

 

IP calls.

 

 

 

 

 

 

 

 

 

 

SIP T1 Timeout

 

 

T1 is an estimate of the round-trip time between the client and server transactions.

 

 

 

 

 

If the network latency is high, select larger value for more reliable usage.

 

 

 

 

 

 

 

 

 

SIP T2 Interval

 

 

Maximum retransmission interval for non-INVITE requests and INVITE responses.

 

 

 

 

 

 

 

 

 

 

DTMF Payload Type

 

 

This parameter sets the payload type for DTMF using RFC2833

 

 

 

 

 

 

 

 

 

 

Preferred DTMF method

 

 

The HT503 supports up to 3 different DTMF methods including in-audio, via RTP

 

 

(in listed order)

 

 

(RFC2833) and via Sip Info. The user can configure DTMF method in a priority list.

 

 

 

 

 

 

 

 

 

Disable DTMF

 

 

Default is No. If set to yes, use above DTMF order without negotiation

 

 

Negotiation

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Send Flash Event

 

 

Default is No. If set to yes, flash will be sent as DTMF event.

 

 

 

 

 

 

 

 

 

 

Enable Call Features

 

 

Default is Yes. (If Yes, call features using star codes will be supported locally)

 

 

 

 

 

 

 

 

 

Offhook Auto-Dial

 

 

This parameter allows users to configure a User ID or extension number to be

 

 

 

 

 

automatically dialed when offhook. Please note that only the user part of a SIP address

 

 

 

 

 

needs to be entered here. The HT503 will automatically append the “@” and the host

 

 

 

 

 

portion of the corresponding SIP address.

 

 

 

 

 

 

Note: User will need this IP address when accessing the IVR via the web configuration

 

 

 

 

 

page.

 

 

 

 

 

 

 

 

 

 

Proxy-Require

 

 

SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.

 

 

 

 

 

 

 

 

 

 

Use NAT IP

 

 

NAT IP address used in SIP/SDP message. Default is blank

 

 

 

 

 

 

 

 

 

 

Distinctive Ring Tone

 

 

Custom Ring Tone 1 to 3 with associate Caller ID: when selected, if Caller ID is

 

 

 

 

 

configured, then the device will ONLY uses this ring tone when the incoming call is

 

 

 

 

 

from the Caller ID. System Ring Tone is used for all other calls. When selected but no

 

 

 

 

 

Caller ID is configured, the selected ring tone will be used for all incoming calls.

 

 

 

 

 

Distinctive ring tones can be configured not only for matching whole number, but also

 

 

 

 

 

for matching prefixes. In this case symbol * (star) will be used.

 

 

 

 

 

 

If server supports Alert-Info header and standard ring tone set (Bellcore) or distinctive

 

 

 

 

 

ring tone 1-10 is specified, then the ring tone in the Alert-Info header from server will be

 

 

 

 

 

used.

 

 

 

 

 

 

For example:

 

 

 

 

 

 

If configured as *617, Ring Tone 1 will be used in case of call arrived from

 

 

 

 

 

Massachusetts. Any other incoming call will ring using cadence defined in parameter

 

 

 

 

 

System Ring Cadence located under Advanced Settings Configuration page.

 

 

 

 

 

 

 

 

 

 

Disable Call Waiting

 

 

Default is No.

 

 

 

 

 

 

 

 

 

 

Disable Call Waiting

 

 

Default is No. This is to disable the caller ID when a call waiting information arrives.

 

 

Caller ID

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Disable Call Waiting

 

 

Default is No. This is to disable the stutter Call Waiting Tone when a Call Waiting

 

 

Tone

 

 

information arrives. The CWCID information will still be displayed.

 

 

 

 

 

 

 

 

 

Disable Reminder Ring

 

 

Default is N

o. The reminder ring for the on-hold call will not be played when this is set

 

 

for On-Hold Call

 

 

to Yes.

 

 

 

 

 

 

 

 

 

 

Disable Visual MWI

 

 

If set to “YES”, the MWI information will not be transferred to the analog phone

 

 

 

 

 

connected to the FXS port.

 

 

 

 

 

 

 

 

 

Ring Timeout

 

 

Sets the time in which an incoming call will stop ringing when not picked up.

 

 

 

 

 

 

 

 

 

 

 

Default value is 20 seconds. In case this feature activated using * codes (*92 code),

 

 

 

 

 

the call will be forwarded after this preconfigured amount of time.

 

 

 

 

 

 

 

 

 

No Key Entry Timeout

 

 

Default is 4 seconds.

 

 

 

 

 

 

 

 

 

Early Dial

 

 

Default is No. Use only if proxy supports 484 response. This parameter controls

 

 

 

 

 

whether the phone will send an early INVITE each time a key is pressed when a user

 

 

 

 

 

dials a number. If set to “Yes”, an INVITE is sent using the dial-number collected thus

 

 

 

 

 

far. Otherwise, no INVITE is sent until the “(Re-)Dial” button is pressed or after about 5

 

 

 

 

 

seconds have elapsed. The “Yes” option should be used ONLY if there is a SIP proxy

 

 

 

 

 

configured and the proxy server supports 484 Incomplete Address response.

 

 

 

 

 

Otherwise, the call will likely be rejected by the proxy (with a 404 Not Found error).

 

 

 

 

 

Note: This feature is NOT designed to work with and should NOT be enabled for direct

 

 

 

 

 

 

 

 

 

 

Grandstream Networks, Inc.

 

 

 

HT503 User Manual

Page 26 of 38

 

 

 

 

 

 

Firmware 1.0.4.2

Last Updated: 06/2011

 

Image 26
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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