Grandstream Networks HT702 SIP Registration, Unregister on Reboot, Outgoing Call without, Contact

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request to indicate the E.164 number. If server supports TEL URI format, then this

 

 

 

 

 

 

 

 

 

 

 

option needs to be selected.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

SIP Registration

 

 

 

Controls whether the HT701 needs to send REGISTER messages to the proxy server.

 

 

 

 

 

 

 

 

 

 

 

The default setting is Yes.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Unregister on Reboot

 

 

 

Default is No. If set to Yes, the SIP user’s registration information will be cleared on

 

 

 

 

 

 

 

 

 

reboot.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Outgoing Call without

 

 

 

Default is No. If set to “Yes,” user can place outgoing calls even when not registered (if

 

 

 

 

 

Registration

 

 

 

allowed by Internet Telephone Service Provider) but is unable to receive incoming

 

 

 

 

 

 

 

 

 

calls.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Register Expiration

 

 

 

This parameter allows the user to specify the time frequency (in minutes) the HT70X

 

 

 

 

 

 

 

 

 

refreshes its registration with the specified registrar. The default interval is 60 minutes

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

(or 1 hour). The maximum interval is 65535 minutes (about 45 days).

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Registration Retry Wait

 

 

 

Retry registration if the process failed. Default is 20 seconds.

 

 

 

 

 

 

 

 

Time

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Local SIP port

 

 

 

Defines the local SIP port the HT70X will listen and transmit. The default value for FXS

 

 

 

 

 

 

 

 

 

port is 5060.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Local RTP port

 

 

 

Defines the local RTP-RTCP port pair the HT70X will listen and transmit. It is the base

 

 

 

 

 

 

 

 

 

RTP port for channel 0. When configured,

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

channel 0 uses this port _value for RTP and the port_value+1 for its RTCP

 

 

 

 

 

 

 

 

 

 

 

The default value for FXS port is 5004.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Use Random Port

 

 

 

Default is No. This parameter forces the random generation of both the local SIP and

 

 

 

 

 

 

 

 

 

RTP ports when set to Yes. This is usually necessary when multiple HT70X are behind

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

the same NAT.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Refer to Use Target

 

 

 

Default is No. If set to YES, then for Attended Transfer, the “Refer-To” header uses the

 

 

 

 

 

 

 

Contact

 

 

 

transferred target’s Contact header information.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Transfer on Conference

 

 

 

Default is No. In which case if the conference originator hangs up the conference will

 

 

 

 

 

 

 

Hang up

 

 

 

be terminated. When option YES is chosen, originator will transfer other parties to

 

 

 

 

 

 

 

 

 

each other so that B and C can choose to either continue the conversation or

 

 

 

 

 

 

 

 

 

 

 

hang up.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Enable Ring-Transfer

 

 

 

Default is No, this will create a Semi-Attendant Transfer. When set to Yes, device can

 

 

 

 

 

 

 

 

 

transfer the call upon receiving ring back tone or SIP message 180.

 

 

 

 

 

 

 

 

 

 

 

Disable Bellcore Style

 

 

 

Default is No. you can make a Conference by pressing ‘Flash’ key. If set to Yes, you

 

 

 

 

 

 

 

3-Way Conference

 

 

 

need to dial *23 + second callee number.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Remove OBP from

 

 

 

Default is No. When option YES is chosen, the Out Bound Proxy will be removed from

 

 

 

 

 

Route Header

 

 

 

Route header.

 

 

 

 

 

 

 

 

 

 

 

 

 

Support SIP Instance ID

 

 

 

Default is Yes. If set to Yes, the contact header in REGISTER request will contain SIP

 

 

 

 

 

 

 

 

 

Instance ID as defined in IETF SIP Outbound draft.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Validate incoming SIP

 

 

 

Default is No. If set to yes all incoming SIP messages will be strictly validated

 

 

 

 

 

message

 

 

 

according to RFC rules. If message will not pass validation process, call will be

 

 

 

 

 

 

 

 

 

rejected.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Check SIP User ID for

 

 

 

Default is No. Check the incoming SIP User ID in Request URI. If they don’t match, the

 

 

 

 

 

incoming INVITE

 

 

 

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

 

 

 

 

 

 

 

 

 

IP calls.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Allow Incoming SIP

 

 

 

Default is No. Check the incoming SIP messages. If they don’t come from the SIP

 

 

 

 

 

Messages from SIP

 

 

 

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

 

 

 

 

 

Proxy Only

 

 

 

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. Default is 0.5

 

 

 

 

 

 

 

 

 

Sec.

 

 

 

 

 

 

 

 

 

 

 

 

 

SIP T2 Interval

 

 

 

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

 

 

 

 

 

 

 

 

 

 

 

Default is 4 Sec.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

DTMF Payload Type

 

 

 

Sets the payload type for DTMF using RFC2833. Default is 101.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Grandstream Networks, Inc.

 

HT-70X User Manual

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Firmware Version 1.0.0.18

Last Updated: 03/2012

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Contents HT701 HT702 Grandstream Networks, IncOnfiguring the HT70X Through V Oice P Rompts Configuration GUI Interface Examples Warranty Safety CompliancesWelcome Connect Your HT70X Connecting the HT70XEquipment Packaging Power LED Definitions of the HT70X ConnectorsBasic Definitions of the HT70X LEDs Pattern HT702 HT704Advanced Definitions of the HT70X LEDs Pattern Analog HT70X Software Features Software Features OverviewProduct Overview HT70X Hardware and Technical Specifications Hardware SpecificationEMC Understanding HT70X Voice Prompt Basic OperationsHT70X IVR Menu Definitions Using Star Code Using IVRPlacing a Phone Call ExamplesCall Waiting Call HoldCall Transfer Or Voice Prompt with option 47, then 192*168*0*160Fax Support Way ConferencingAttended Transfer Instructions for 3-way conferenceHT70X Call Features Call FeaturesEnable Srtp Disable Srtp Blind TransferConfiguring the HT70X through Voice Prompts Configuration GuideConfiguring the HT70X Via Web Browser NAT Settings Important SettingsBasic Settings MTZ+6MDT+5 StatusNAT Advanced Settings Advanced User ConfigurationDND FXSHTTP/HTTPS URI ACS URLPrimary Radius Syslog LevelAccount Settings Enable Ring-Transfer Unregister on RebootDisable Bellcore Style SIP T1 TimeoutEnable Call Features Disable DtmfDisable Call Waiting Disable Call-WaitingEarly Dial No Key Entry TimeoutDial Plan Prefix Use # as Dial KeyVAD Detection Mode Fax ModeSrtp Mode Slic SettingSaving the Configuration Changes HT704 FXS Ports SettingsRebooting the HT70X from Remote Configuration through a Central Server Firmware Upgrade through TFTP/HTTP/HTTPS Software UpgradeInstructions for Upload from Local Directory Configuration File Download Instructions for local firmware upgrade using Tftp serverFirmware and Configuration File Prefix and Postfix Managing Firmware and Configuration File DownloadYes, every Encode the MAC Address Restore Factory Default Setting
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HT704, HT701, HT702 specifications

Grandstream Networks has established itself as a powerful player in the telecommunications industry, especially through its Analog Telephone Adapters (ATAs) like the HT702 and HT704 models. These devices are designed specifically for converting analog voice signals into digital data for seamless integration with Voice over Internet Protocol (VoIP) systems.

The Grandstream HT702 is a two-port ATA that allows users to connect two analog phones to a high-speed internet connection. This model is particularly useful for small businesses or residential users looking to integrate legacy phone systems with modern VoIP technology. One of the key features of the HT702 is its support for the SIP (Session Initiation Protocol) standard, ensuring compatibility with a wide range of VoIP providers. Additionally, it supports advanced telephony features like call transfer, call waiting, and three-way calling, enhancing communication efficiency.

The HT704, on the other hand, is a four-port ATA, offering greater flexibility for users needing to connect multiple devices. It shares many of the same features as the HT702, including SIP support and telephony functionalities, but with additional ports, it is better suited for larger environments. Both models come equipped with advanced security mechanisms, such as AES encryption, which safeguards voice communications.

With user-friendly web-based configuration, the HT702 and HT704 allow for easy setup and management, making them accessible even for those without extensive technical knowledge. Moreover, both devices feature auto-provisioning capabilities, which simplify deployment across multiple units, making them ideal for businesses looking to scale their operations.

The HT702 and HT704 are built with high-quality materials, ensuring durability and long-term performance. They also boast low power consumption, making them an energy-efficient choice. Support for high-definition voice codecs enhances audio quality during calls, providing users with crystal-clear communication.

In summary, Grandstream's HT702 and HT704 Analog Telephone Adapters are robust solutions for anyone looking to transition from traditional telephony to a modern VoIP setup. Their advanced features, security standards, and ease of use make them a reliable choice for both home and business users seeking efficient and effective communication solutions.