Grandstream Networks GXW410X Wait for Dial­tone, Stage Method, Min. Delay before, Dial Pstn

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Wait for Dial­tone

Default is No. When set to Yes, the gateway will recognize dial­tone from the Central Office (CO)

 

before it completes a call. If you can’t make an outbound call, set this is Yes. if this is set to Yes

 

make sure you have configured the dial tone settings correctly in the Channels tab and that there

 

is not any major noise interference in the line.

Stage Method

Syntax ­ ch1­8:1; {all channels 1 to 8 are set to value 1 or 2}

 

Stage method can be set to either 1 or 2.

 

Set this parameter to 1 if you need to make a direct PSTN call from a VOIP endpoint. When you

 

set it to 2, you will first dial one of the VOIP channel accounts from the VOIP endpoint, this will

 

result in getting a PSTN line dial­tone to then dial out the destination PSTN number.

 

Most implementations require this setting to be configured to 1.

Min. Delay before

Default is 500ms. This needs to be equal to or greater than the Current Disconnect threshold

Dial PSTN

setting. Once the threshold is reached the gateway can dial out. This parameter should only be

 

used if there are PSTN line detection issues.

Unconditional Call

This is an extremely important setting to make sure incoming PSTN calls are picked up and

Forward to VOIP:

forwarded to the correct VOIP destination.

 

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

 

automatically dialed upon FXO line off­hook.

 

SIP Server ­ You also need to specify the Profile of the user id configured above (p1 stands for

 

Profile 1, p2 stands for Profile 2 and so on).

 

SIP Destination Port ­ Along with the user­id and Profile, you also have the option to choose the

 

destination port where you would like to send the call. By default it should be set to ch1­x:5060; (x

 

can be 4 or 8 depending on number of ports).

 

We can also specify a different destination for each port. For example under User ID we can type

 

in: ch1:104;ch2:227;ch3­5:501;ch6,7:856.

 

Under Sip Server we can type in: ch1:p1;ch2­4:p2;ch5:p3

 

Under Sip Destination Port we can type in: ch1­2:5060;ch2:7080;ch3­8:5066++

Number of Rings

Default is 4. This is the number of rings the gateway will wait to send the call to the

Before Pickup

VOIP side in case the Caller ID has yet to be detected. If there's CID information the

 

call will be sent right away. If your lines don't have the CID service set this to 1.

 

 

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Contents Grandstream Networks, Inc Table of Contents Table of Figures GUI Interfaces Welcome LAN or PC PackagingReady LED Power LEDVideo LED LEDs 1 ­Pstn Analog endpoints Grandstream IP Phones Anywhere WorldLines Ports 410xFXS Gateway with GXW410x No SIP Server required Channels LEDs Green LAN interface 2xRJ45 10/100MbpsLED Indicators IP settingsOn/Off Switch Pstn SignalingUser Level Password Web pages allowed Configuration GuideEnd­user Password Web AccessPPPoE password PPPoE Service NameHardware Revision Daylight Savings TimeMAC Address Product ModelFXO Line Connected PPPoE Link UpDetected NAT Type Password AdminG723 Rate Voice FramesPath Config Server UpgradePostfix Config File Prefix Config FileDtmf Payload Option 42 to Override an NTP Server Enable VideoType Syslog Server Syslog LevelEnable Current Syntax for Channel Specific SettingsEnable Tone Enable PolarityWait for Dial­tone Min. Delay beforeStage Method Dial PstnCaller ID Scheme SettingCaller ID Transport TypeLine # External Call TimeoutAC Impedance CPT DetectionAuthentication ID Error TimeoutAuthentication Password Channel Voice SettingsChannel specific Setting Echo CancellationNo Key Entry Timeout Srtp ModeHook flash Duration X10ms Use Dtmf ParameterFrom RFC2833 or SIP Info Timer Reboot Register ExpirationSIP Server Outbound ProxyEnable 100rel Force TimerAnswer Timeout Special FeatureVideo Surveillance  Gateway side∙ Deviceipaddress is the device IP PC client side running VLC as monitoring station Firmware Upgrade Directions for Downloading Tftp Server Restore Factory Default Setting Pstn Analog Lines FXO Lines Ports410x Grandstream IP Phones Cloud Internet CloudPstn
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GXW410X specifications

Grandstream Networks, a leading provider of IP voice and video communication solutions, has made a significant mark in the telecommunications industry with its GXW410X series of Analog Gateway devices. These gateways serve as a bridge between traditional analog phone systems and modern IP networks, making them an ideal solution for businesses looking to enhance communication efficiency while leveraging existing infrastructure.

At the heart of the GXW410X series is a range of models designed to support different analog port configurations, including the GXW4104, GXW4108, and GXW4116, which provide 4, 8, and 16 FXS ports, respectively. This flexibility allows organizations to choose the right model that fits their specific needs, whether they are a small business or a larger enterprise.

One of the standout features of the GXW410X is its robust SIP (Session Initiation Protocol) support, which facilitates easy connectivity to various VoIP service providers. This capability ensures that users can make and receive VoIP calls over their existing analog lines, reducing operational costs while improving call quality. The device also supports up to 1000 SIP accounts, allowing multiple users to make simultaneous calls without compromising performance.

The GXW410X series provides advanced call management features, including caller ID, call waiting, call forwarding, and voicemail, which empowers users to enhance their communication experience. Additionally, it offers integrated NAT (Network Address Translation) support and security features such as built-in firewall and encryption protocols, ensuring secure operations within the IP network.

The scalability of the GXW410X series is another remarkable characteristic. It enables seamless expansion for businesses as they grow, with the ability to easily connect additional analog devices to the network without significant infrastructure changes. Furthermore, the installation and configuration processes are user-friendly, thanks to the intuitive web-based management interface that simplifies monitoring and administration tasks.

Power over Ethernet (PoE) support is also included in the GXW410X series, allowing devices to receive power via the same Ethernet cable used for data transmission. This feature reduces the need for additional power adapters and simplifies deployment in various environments.

In summary, Grandstream Networks' GXW410X series of Analog Gateways combines sophisticated technology with practical features to deliver a reliable and cost-effective communication solution. With its SIP support, extensive call management options, security measures, and scalability, the GXW410X is an excellent choice for organizations looking to modernize their telephony systems while optimizing their resources. This device epitomizes Grandstream's commitment to quality, reliability, and innovation in the telecommunications space.