Ejoin VoIP And SMS Gateway Modem Advanced Setting
Network settings
1.PPTP-VPN settings
A virtual private network (VPN) extends a private network across a public network, such as the Internet. It enables a computer or network-enabled device to send and receive data across shared or public networks as if it were directly connected to the private network, while benefiting from the functionality, security and management policies of the private network. This device works as VPN(PPTP) client mode only, if you want to use VPN function, please input the VPN parameter on the PPTP-VPN settings page.
Ejoin VoIP and SMS gateway PPTP-VPN Settings
2.Network Settings
There are three ways to access the device: web, telnet and serial. web default port is 80, telnet is 23 and serial is the com port you insert. Web configuration is widely used in this device. If you want to change web and telnet default port, please input new port on this page.
Ejoin VoIP and SMS gateway Network Management Settings
Port Settings
Ejoin VoIP and SMS gateway Port Settings
| Items | Description |
| Type | Indicates the current type of network GSM/CDMA/WCDMA. |
| Disable | If it is disabled, this channel will be locked by gateway. |
| Hot-line | When GSM part client call to this channel, gateway will auto forward to the hot-line (Mobile to VoIP). Leave it blank if you don’t need this function. |
| Unconditional Forward | When GSM part client call to this channel, gateway will forward the call to another mobile unconditionally. |
| No Answer Forward | When GSM part client calls to this channel, if this channel is no answer, gateway will forward the call to another mobile. |
| Busy Forward | When GSM part client call to this channel, if this channel is busy, gateway will forward the call to another mobile. |
Other Settings
Ejoin VoIP and SMS gateway Application Feature
| Items | Description |
| Caller ID Display | If it is disabled, caller ID will not show on “call status” page. |
| Silence Suppression | If it is enabled, half of the bandwidth will be saved. |
| Adaptive Jitter Buffer | A jitter buffer is a shared data area where voice packets can be collected, stored, and sent to the voice processor in evenly spaced intervals. |
| IP TOS | TOS of IP packets. |
| Don’t send # to PSTN | If it is enabled, the last digit # of callee number will be removed. |
| Append # to PSTN | If it is enabled, # will be appended in the callee number |
| Forbid GSM call | Calls will be rejected when calling the SIM in gateway. |
| White Number List | The numbers in white list will not be rejected if forbid GSM call is enabled. |
| DTMF Pre-Act time | The prepare time until DTMF tone is detected. |
| DTMF Activity time | The minimum of DTMF activity time. |
| Max Alerting Time | The maximum time of alerting. |
| Max Ringback Time | The maximum time of ring back. |
| Max Call Duration | The maximum duration for every call. System will hang up the call automatically if the call duration excesses this value. |
| RTP Inactivity Time | The maximum duration of silence from gateway. System will hang up the call automatically if the silence duration excesses this value |
| Auto Alerting Time | Fake ring back time, gateway will do fake ring back when excesses this value. |
| Stop Pseudo Time | Stopping fake ring back when the callee is alerting. |
| GSM Auto Answer | Applying to calls from GSM network. The gateway will answer the incoming calls automatically when excesses the value. |
| VoIP Call Auto Answer | Applying to calls from IP network. The gateway will answer the calls automatically when excesses the value. |
| DTMF Mode | RFC2833, SIP INFO and IN-BAND. The default one is RFC2833. |
| RFC2833 Payload Type | RTP Payload for DTMF, the default is 101. |
| RTP Ptime | The interval of RTP packages. |
| RTP Start Port | The initial port when RTP voice stream transmit the IP network. |