• How to handle incoming calls on VoIP GSM Gateway ?
    By default, when there is an incoming call on VoIP GSM Gateway, if user does not have any configurations, the gateway will play default IVR “please dial extension”. If users want to set or disable that, please refer user manual or seek help from support team.
    What is the concept of Ringback tone of UC2000 VoIP GSM Gateway
    Ringback tone is an audible indication that is heard by the originator of a telephone call while the destination being calling is ringing. It is normally a repeated tone, designed to assure the calling party's line is ringing.
    How To check SIM Card balance?
    The customer want check balance auto and block simcard when no_balance.The UC2000 have 3 modes to check balance: USSD/SMS/Call.
    How to Learn SIM Card Number？
    The customer want learn the SIM card number and used for auto call. The UC2000 have 3 modes to learn SIM card number: USSD/SMS/Call.
    How to configure abnormal call handle？
    This document is trying to explain the function of “Abnormal Call Handle” on UC2000 gateway. The document contains all necessary elements required to use the abnormal call handle properly. For any further support, please contact support@dinstar.com.
    What is the concept of NAT?
    "Network Address Translation" is a method for translating the (mostly private) IP addresses of a network onto other (mostly public) IP addresses of another network. NAT therefore enables several PCs in an LAN on the one hand to use the IP address of the Internet Access Router for Internet access and on the other it hides the LAN behind the IP address of the router registered on the Internet. NAT therefore spares the need for each user to have a separate provider contract. If, then, the client in the LAN sends an IP packet to the router, NAT converts the address of the sender into a valid IP address, which for instance has been assigned to it by the provider, before it is passed on onto the Internet. If an answer to this packet comes back from the remote station, the NAT converts the receiver address back into the original IP address of the local station and delivers the packet in proper form. In theory, NAT can manage LANs with any number of clients.
  • What is the default IP address of Analog VoIP Gateway?
    The default IP address at LAN port is 192.168.11.1. The WAN port IP is DHCP. You can change IP address when login the device configuration page.
    Why the web browser can not access the gateway default IP adress-192.168.11.1?
    Please be make sure your PC had added IP at same range of 192.168.11.x/24 already, like 192.168.11.2/24.
    What are the default username and password of Analog VoIP Gateway?
    The default username is 'admin'. The default password is also 'admin'. You can change username and password on configuration page.
    How can I know the IP address of Analog VoIP Gateway?

    For FXS gateways, connect an analog phone to the FXS port, dial the number *158#, and you will hear the LAN port IP.

    For FXO gateways, connect a PSTN line to the FXO port, dial the PSTN line number from your mobile phone, press '*158#' when you hear the IVR, then dial the extension number, and you will hear the LAN port IP.

    Note: If the FXO gateway is set to auto dial, this method will not work.

    How can I restore the device to factory default settings?
    keep the Access Gateway UP. Press the RST button 5s till device RUN LED status changed Reboot the Access Gateway
    How to make sure which correct Caller ID detection mode use: Before ringing or After ringing?
    Firstly, you should connect the PSTN line to an analog phone directly. Then make an incoming call (using another line or mobile phone) to that PTSN line and check analog phone displays Caller ID. If the Caller ID is not shown on the phone, means the CID disable at Telco side. please contact your telco or PSTN service provider. If the analog phone shows the correct CID, please take note if the Caller ID shows before the first ring tone or after the first ring tone: - Before ringing indicates that the PSTN line will send Caller ID before the first ring signal. - After ringing indicates that the PSTN line will send Caller ID between the first ring and second ring.
    How to check the Analog Media Gateway if call out failed?
    Firstly, please check can get dial tone or not when pickup analog phone. If not, need check the FXS port registered or not at device page “Status & Statistics”. If FXS port registered already, confirm the route setting correct at page “Tel->IP”. After check above and issue still, try to capture the call log though page “Network Capture” , send log to support@dinstar.com
  • What is the default IP address of Digital VoIP Gateway?
    The Digital VoIP Gateway have two Ethernet port. One for Service, one for Management. MTG200/600/1000/ 1000B: FE0 (Service port), IP: 192.168.1.111 FE1 (Management port), IP: 192.168.11.1. MTG2000/3000: GE1 (Service port), IP: 192.168.1.111 GE0 (Management port), IP: 192.168.11.1
    Why my web browser can’t open 192.168.11.1 or 192.168.1.111?
    Please make sure you PC connected which type Ethernet port at MTG. Then, added the corresponding IP at PC: 192.168.11.x/24 or 192.168.1.x/24. Like 192.168.11.2/24
    What is the default username and password of Digital VoIP Gateway?
    The default username and password are : Username:admin Password:admin You can change username and password on configuration page.
    How can I restore the device to factory default settings?
    Login the Digital VoIP Gateway. Open the page “Management>>>Data Restore”, upload the blank database file. Reboot the Digital VoIP Gateway
    How can i restore the login information?
    Keep the device ON. Press the RST button still RUN LED status changed. Reboot the device. After device UP, the Management port setting will back to default. This option not affect the Service port setting. Note: MTG1000 not support this one.
    How to fix it if face “ISDN/SS7 Signal Alarm”?
    Please check the E1 cable used Wire Sequence. The E1 Wire Sequence are different with Net cable. It use 1,2,4,5: 1: Rx Ring- 2: Rx Tip+ 4: Tx Ring- 5: Tx Tip + If the cable OK, need capture log and send to support@dinstar.com Please see the Q9/Q10, it show you how to do.
    How to check the Digital VoIP Gateway if call no voice?
    Firstly, make sure the IP trunk bind at Service port. If bind at Management port, will face voice issue. Second, check the MTG side used RTP port range, the range don’t include sip port, like 5060t. Recommend use 8000 or above. If use SS7, please check the CIC set and confirm it correct.
    How to get PRI log of Digital VoIP Gateway?
    Telnet of device and input follow command : ROS>en ROS#sh version ROS#sh int ROS#sh mcc X//X: Port0 is 16, Port1 is 18, port2 is 20 //Do the same commend “sh mcc x” each 10s, 5 times total. ROS#^config ROS(config)#deb Q931 detail       ROS(config)#ex ROS#^ada ROS(ada)#turnon 64
    How to get SS7 log of Digital VoIP Gateway?
    Telnet of device and input follow command : ROS>en ROS#sh version ROS#sh int ROS#sh ss7 error clear            //clear the Error log  ROS#sh SS7 link                     //show current SS7 hdlcNo and LinkID, we will use it at follow ROS#sh mcc X                       //Check the error frame, x means hdlcNo x //Do the same commend “sh mcc x” each 10s, 5 times total.    ROS#sh ss7 error                 //show the Error log  again ROS#^config ROS(config)#deb ss7 Y 5      // The Y means the SS7 LinkID index Y ROS(config)#ex ROS#^ada ROS(ada)#turnon 96
  • How to install antenna of UC120 ?
    There are two kinds of antennas in UC120, one is for Wi-Fi, the other is for GSM/LTE. Wi-Fi antennas are round which should be installed to the left and right side of the gateway. GSM/LTE antennas are flat, which should be installed to the back panel of the gateway.
    What is the default IP address of IP PBX?
    The default IP address at LAN port is 192.168.11.1. The WAN port IP is DHCP. You can change IP address when login the device configuration page.
    What are the default username and password of IP PBX?
    The default username and password of UC100 are : Username:admin Password:admin The default username and password of UC200 are : Username:admin Password:admin@123# You can change username and password on configuration page.
    How to use Wi-Fi of UC120?
    If your model does not support VoLTE, you can only use WAN mode, which requires connecting the WAN port to the network to access Wi-Fi data. However, if your model supports VoLTE, inserting a VoLTE SIM card will allow you to access Wi-Fi directly through the SIM card. Additionally, when you connect the WAN port to the network, you can access Wi-Fi through the WAN connection. By default, the strategy is set as WAN Master, LTE Slave. However, you can modify this strategy by navigating to the "Network" page and selecting "Uplink Control."
    How can I restore the device to factory default settings?
    On the condition that the device is running normally, press the RST button for 6-12 seconds, and all configuration are restored to the default settings.
    How to trace logs of IP PBX?

    To debug calls or voice problems, you can go to web page 'Call Control->Diagnostics' page. And select debug logs whatever you want.

    SIP Stack: logs of SIP stack.

    SIP Message: logs of SIP message like Register, INVITE request etc.

    FXS/FXO: logs of FXS or FXO port

    GSM: logs of GSM modules

    DSP: capture PCM packets.

    Voice: SIP and RTP capture, you can open the capture by Wireshark program.

    How to understand Number Profile and Manipulation?
    There are two import profiles which can be used in Route. One is Number profile, the other is Manipuliation. Profile -> Number profile, it is a kind of number condition, once you create and apply it in Route, it means the number which matched the number profile can be allowed; Profile -> Manipulation, it is a way to change caller or callee number, when you apply it in Route, it means it will change the number before sending to the corresponding destination.
    How to configure Call Transfer from IPPBX?
    Call Transfer including blind transfer and attended transfer. You can check the feature code of transfer in page Call Control -> Feature Code. Blind transfer is a call transfer in which the transferring party connects the call to a third party without notifying the third party. Attended transfer is a call transfer in which the transferring party connects the call to a third party after he confirms that the third party agrees to answer the call.
    How many types of Call Forwarding?
    In general, there are 4 types of call forwarding. Call forwarding service includes unconditional call forwarding, unregistered call forwarding, call forwarding on busy, call forwarding on no reply.