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fe 3000 gsm manualTo start viewing messages,I now have a programming scenario that I'm sure it can do quite easily it's just I have only programmed the basics of these and I feel I now have to delve into different routes My problem is this. We have a client that has a chain of shops which all have GSM's fitted. Most of them are other types but one in particular has an FE 3000 fitted. This has been done at the exchange so no wiring has changed onsite. Suddenly overnight all of their stores stopped reporting and we have had to visit each (over 20) to change the receiver numbers and test. The FE 3000's are different from most other GSM;s in that its the GSM unit itself which kisses off the panel, and then re-transmits the data to the control room. It also stores the phone number from the panel (on its first received signal) as well as the account number. The unit uses this number to dial the control room whether its the pstn line it uses or the GSM route. If we remove the '0' then the pstn has the wrong number but the Gsm gets through. How can I tell the Gsm to use the receiver number withouth the '0' when its reporting via gsm network but to use the '0' when using the pstn network. Hope this all makes sense ThanksDo you programme your GSM's to send a timer test or anything ? (i know its un related, but i find so many gsm's with no timer test and no gsm fail output to the panel, drives me crazy)If the panel number is un-programmed (default), then it will be replaced by the 1st phone number used by the alarm panel on its first communication to the FE3000.If you don't have a manual PM me for one.Do you programme your GSM's to send a timer test or anything ? (i know its un related, but i find so many gsm's with no timer test and no gsm fail output to the panel, drives me crazy) If you don't have a manual PM me for one. I will try this on a bench before going to site. Sounds like a plan though. Also, I do have a manual so will give it a shot.http://www.salmododia.com.br/imagens/imagens_usuarios/engineering-procedure-manual.xml
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Thought it might be a bit more complicated than that by have to change the destinations etc. Also was wondering what the pabx option on the FE 3000 does. Not sure if its implemented in all versions though. PaulThe FE3000 will use PN (Panel Number) as the receiver number for all communications, unless you program in the TN numbers. If the FE has the basic default config, TN1 and TN2 will be dialled via PSTN, TN3 and TN4 will be dialled via GSM. Reload to refresh your session. Reload to refresh your session. You can always update your selection by clicking Cookie Preferences at the bottom of the page. It may also provide information about features and functions, applications and troubleshooting. By downloading manuals from Tektronix' website, you agree to the following terms and conditions: Manuals for currently supported products may not be reproduced for distribution to others unless specifically authorized in writing by Tektronix, Inc. Thus, different versions of a manual may exist for any given product. Care should be taken to ensure that one obtains the proper manual version for a specific product serial number. Tektronix hereby grants permission and license for others to reproduce and distribute copies of any Tektronix measurement product manual, including user manuals, operator's manuals, service manuals, and the like, that (a) have a Tektronix Part Number and (b) are for a measurement product that is no longer supported by Tektronix. Thus, different versions of a manual may exist for any given product. Care should be taken to ensure that one obtains the proper manual version for a specific product serial number. ME3000V2 Operation Description. Version:V1.0Table of Contents. Mechanical Interface. 3Antenna Pad. 7. Antenna connector. 7. RF Interface. 9Antenna Interface. 7Power Management. 6Mechanical size. 3. Overview. 9. Antenna Subsystem. 9Test Capabilities. 12GSM Test Equipment and Tools. 13Operational. Temperature Range. Storage Temperature.https://alexis-services.com/userfiles/engineering-processes-manual.xml RangePower Supply. Length. Width. Thickness. Weight:YesTwo kinds of inputs:Figure 1-1: ME3000V2 T-viewgraphFigure 2-1 ME3000V2 Module interface ME3000V2 map. Table 2-1 ME3000V2 Module 40-pin Electrical Interface. Pin. Signeal. Name. Signal. Type. InpuMin. Typ. Max. Uni. CommentsSIM clockSIM powerDigital. Digital. Power. SIM data. SIM reset. LED control. COM Port. Main powerDigital. DigitalCOM Port. COM PortCOM PortCOM PortCOM PortSystem resetAnalog. Analog. AnalogMic inputSpeakerLED ON asHeadset MIC. EarphoneReceiver. Earphone. ReceiverThe module could work under two power modes: 1. Charger; 2 Battery. When powered by the charger, you could perform constant current charge, constant voltageNormally, trickle current charge starts when the voltage isSee table 4-1 for the module’s input voltage characteristics. If the input voltage is not in theTable4-1 Voltage CharacteristicsStatus. Max. voltage. Power supply. Typical voltage. Min. voltageStatus. Power supplyTypical voltage. Min. voltage. Power on. The module is under power-off status after it’s normally powered on. To turn on the module,Use the above method to firstly “Power off” and then “Power on”, and by doing so the moduleThe RF interface of the ME3000V2 Module has an impedance of 50. The module is capable of sustaining aThe external antenna must be matched properly to achieve best performance regarding radiated power. DC-power consumption, modulation accuracy and harmonic suppression. Antenna matching networks are notRegarding the return loss, the Module provides the following values in the active band. Table 4-1 Return Loss in the Active Band. State of Module. Return Loss of Module. Recommended Return Loss of. Application. Receive. TransmitThe connection of the antenna or other equipment must be de coupled from DC voltage. This is necessaryTo suit the physical design of individual applications, the ME3000V2 offers two alternative approached toSee Section 4.3 for details. See Section 4.2 for details.http://www.statcardsports.com/node/11520 The MM9329-2700B connector has been chosen as antenna reference point (ARP) for the ZTEMT referenceAll RF data specified throughout this manual areNote: Both solutions can be applied alternatively. This means,if the antenna is connected to the pad, then theThe antenna pad of the module is soldered to the board on the customer design to connect with RF line. For proper grounding connect the RF line to the ground plane on the bottom of the MG2636 Module whichConsider that according to GSM recommendations as 50.Material PropertiesThe ME3000V2 Module uses a microwave coaxial connector supplied by Murata Ltd. The product name is. MM9329-2700B. The position of the antenna connector on the Module PCB can be seen in Figure 4-1.Figure 4-1 Specification of MM9329-2700B connector. Table 4-2 Product specifications of MM9329-2700B connector. Part. Number. Rated. Voltag. Contact. ResistanWithstandiVoltageInsulatio. ResistanDurabiliFrequenRating. TemperatuRangeContaOuter. ContaCopper. Alloy. Copper. GoldSilverInsulatorImpedance: 50 ohm. EngineeriA 50 ohm coaxial RF connector is provided for Module testing. However, we advise customers lead from theFigure 5-1 GSM Connector. The module must provide a suitable antenna that works in the desired frequency band of operation. The AntennaBand. TX Frequency. RX FrequencyGSM system).The antenna sub-system and its design is a major part of the final product integration. Special attention and careAll cables have RF losses. Minimizing the length of the cable between the antenna and the RF connectors on theContact the antenna vendor for the specific type of cable that interfaces withTypically, theThough the system will work with longerCare should also be taken to ensure that the cable end. Module sub-system. This is particularly important for applications where the module is mounted on a mobile orIt is recommended that the antenna chosen have at least 2 dBi gain in the GSM900 band and 4 dBi in the PCSOur FCC Grant imposes a maximum gain for the antenna subsystem: 7 dBi for the GSM900 band and 13dBi forWarning: Excessive gain could damage sensitive RF circuits and void the warranty.The module’s RF connectors are designed to work with a 50-ohm subsystem. It is assumed that the antennaIt should be isolated as muchSignals like charger. Please contact the antenna vendor for matching requirements.It is essential to keep the voltage ripple to a minimum at this connection in order to avoid phase error. InsufficientEMC performance, and spurious emissions and frequency error. The RF connections are 50-ohm impedance systems and are a DC short to ground. Best effort should be made toOn terminals including the antenna, poor shielding can dramatically affect the sensitivity of the terminal.MoreoverConnect the sector to access terminal antenna connectors as shown in the following figure 6-1Lease or purchase of test equipment is available from vendors who provide this equipment for GSM over the-airRF Performance RequirementsFrequency range. Rx. SensitivityRx. Signal RangeMax. frequency tolerance. Max. Tx. Power. Peak Phase Error. RMS Phase ErrorPDF Version: 1.5. Linearized: Yes. Author: Administrator. Creator: Administrator. Title: Microsoft Word - Q78-ME3000V2 User Manual. Creator Tool: PScript5.dll Version 5.2.2. Producer: Acrobat Distiller 9.0.0 (Windows). Document ID: uuid:8928fa21-931e-46db-a3ce-026358293352. Instance ID: uuid:0e394d57-dd9e-4014-b4a6-e8567599f24f. Page Count: 15. Model: GoIP1.Reset Press this button to reset the GoIP1 Gateway to factory defaults. 6 One-Channel GSM VoIP Gateway 2 Installation 2.1 Installation Steps Please follow the connection diagram above to install the GoIP1 Gateway.Connect an Ethernet cable the LAN port of the GoIP1 Gateway and the other end to your existing network equipment. (Optional) Connect an Ethernet cable to the PC Port of the GoIP1 Gateway and the other end to a PC or other network device. Connect the power adapter provided to the power jack of the GoIP1 Gateway. 7 One-Channel GSM VoIP Gateway 2.2 LED Indicators The following table defines the status of the LEDs located on the top case and on the RJ-45 connectors. LED DESCRIPTION RUN 1. When the GoIP1 is booting, this LED will flash 100ms ON and 100ms OFF. 2. When the GoIP1 is properly registered to your softswitch, this LED flashes at a rate of 1s ON and 1s OFF. GSM When the GSM channel is ready to sue, this LED flashes at a rate of 1s ON and 1s OFF. 2.3 SMS Commands GoIP1 supports some maintenance commands from SMS. FUNCTION SMS CONTENT REMARK Obtain LAN Port Info INFO Not case-sensitive Reset device RESET Password Not case-sensitive Reboot device REBOOT Password Not case-sensitive: Note In command Reset and Reboot, the Password is the GoIP1 device’s admin password. The LAN port is factory preset to IP address 192.168.0.100 and the PC port is set to the fixed IP 192.168.8.1. If you lose the IP address information for LAN port, just dial a call to GoIP1 Gateway’s SIM card phone number. When the call is connected, you will hear a dial tone. The LAN IP Address can also be obtained by sending a SMS message to the GSM phone number. The GoIP1 will then reply with a SMS message containing the LAN IP address. If you want to obtain LAN port IP by sending a SMS message, please send” INFO “or” info”. Another way to access the GoIP1 Gateway is via the PC port. You will need to change your computer’s LAN configuration via the Network Connections under the Control Panel. Once either the IP address of the LAN or PC port is known, you are now ready to access the Web server of GoIP1 Gateway. 9 One-Channel GSM VoIP Gateway 3.1 Web Configuration Menu If your computer is connected to the GoIP1 Gateway via the LAN port, you need to type the LAN IP address of the GoIP1 Gateway in your Web Browser to access the Web server of the GoIP1 Gateway. The default IP address on the LAN port is “192.168.0.100”. If your computer is connected to the GoIP1 Gateway via the PC port, you should type GoIP1’s PC port IP address (192.168.8.1) in the Web Browser. If the connection is correct, the Web Browser will prompt you to enter the “User name” and “Password” as shown below. Enter the User name and Password and the press OK to access the GoIP1 Gateway Web Server. This number is important for centralized configuration, technical support, and warranty. This number is printed on the bottom of the Gateway and is associated with your software license. B. Firmware Version Firmware version identifies the firmware version of the Gateway such as GHS-3.01-36. C. Hardware Mode This field shows terminal’s hardware type. D. Phone Status This field shows the status of Line’s connection status. If the connection is successful, this field displays LOGIN; otherwise, it displays LOGOUT. 3.2.2 Network Information A. LAN Port Configuration This field displays the status of the LAN port. B. PC Port Configuration This field displays the status of the LAN port. C. PPPoE If PPPoE is enabled, it displays its status. D. Default Route This field displays the IP address of the default routing Gateway. E. DNS Server This field displays the IP address of the Domain Name Server. 3.2.3 GSM Module Information A. GSM Module This field displays the GSM module type. B. GSM Signal This field displays the GSM signal status. The value of GSM signal strength RSSI (Received Signal Strength Indication) is between 0 dbm and 31 dbm. The value of 99 means unknown or undetectable. C. GSM Status This field shows the status of GSM connection status. If the connection is successful, this 12 One-Channel GSM VoIP Gateway field displays LOGIN; otherwise, it displays LOGOUT. 3.3 Configurations Click on the “Configurations” tab on the left hand column to access the device configuration menu: Preference, Network, Call Settings, Call Divert, Save Changes, and Discard Changes. Click on “Preference” in the left menu of the configuration web, and the screen will be displayed as below: 13 One-Channel GSM VoIP Gateway 3.3.1 Language Currently GoIP1 only supports English. VADcore also has other versions of software that support Simplified Chinese and Traditional Chinese. Contact VADcore if you need other language support. 3.3.2 Time Zone and Time Server The GoIP1 Gateway supports Network Time Protocol (NTP) to obtain the date and time information from an NTP server (Time Server). For example, the Pacific Standard Time is GMT-8, and the Pacific Daylight Time is GMT-7. Note: The GoIP1 Gateway supports CDR and Billing Information, it is important to set up these two parameters properly. 3.3.3 DTMF Min Detect Time Gap This parameter is used to limit two same DTMF digit’s minimum time gap, the range is 60ms to 120ms, default is 80ms. If you encounter double digit problem, increase this parameter. If you encounter lose digit, then decrease this parameter. 3.3.4 Network Tone Network Tones are a set of tones used for VoIP calls. Select one of the predefined countries 14 One-Channel GSM VoIP Gateway or select “Customized” to define your own Network Tones. The Group Mode works like a multi-channels GSM gateway. Any GoIP1’s channel can work as Group Server Mode or Client Mode. Server Mode: Only one GSM channel runs in Server Mode. The GSM channel that is set in Server Mode will forward the GSM’s incoming calls to other available client channels. The GSM channel that is set in Server Mode will be your main number for your customer. Client Mode: Other GSM channels will run in Client Mode. The GSM channels that are set in Client Mode will report their status to GSM channel that is set in Server Mode. The GSM channel in Server Mode then forwards phone calls to available GSM channels in Client Mode. You must enter the GSM number for that GSM channel and IP address of the device in Server Mode into the field. Disable: Please set all channels to Disable Mode if you would like to run each channel 16 One-Channel GSM VoIP Gateway independently. 3.3.6 GSM Caller ID Anonymous Some GSM ISPs allow the caller to disable the phone number (caller ID) when making outgoing calls. This feature must be supported by GSM ISPs. 3.3.7 GSM Band GoIP1 Supported quad GSM bands: 850MHz, 900MHz, 1800MHz, 1900MHz. Select the correct GSM bands that are used in your country. 3.3.8 SMS Sender VADcore offers a software to send out SMS to GSM network through GoIP1 Gateway. A SMS server is required to work with GoIP1 Gateway for SMS Sender. Please contact VADcore for more details. 3.4 Call Settings Click on the “Call Settings” to configure the VoIP call settings. 17 One-Channel GSM VoIP Gateway 3.4.1 SIP Standard Supported GoIP1 supports SIP standard. GoIP1 has two types of config modes for SIP protocol; Single Server Mode: The channel uses a SIP account to connect to SIP server. Trunk Gateway Mode: The GoIP1 will act as a SIP proxy. Remote SIP clients can register to GoIP1 and GoIP1 will process SIP requests on behalf of SIP client. GoIP1 Gateway’s SIP configure page as follow: )Phone Number Enter a SIP phone number. B)Display Name A Enter this field for the name to be displayed on the called VoIP party. C) SIP Proxy 18 One-Channel GSM VoIP Gateway Enter the SIP proxy IP address or domain name. If the registration port is not 5060, then add “:” and the port number. For example: sip.hybertone.com:8080. SIP Registrar Server Enter the SIP registrar server IP address or domain name in this field. If the registration port isn’t 5060, add “:” and the port number. For example: sip.hybertone.com:8080. Register Expiry(s) Enter the register time (seconds) in this field. This is the maximum length of registration that SIP server will keep your registration. If SIP server does not receive another SIP registration, the current registration will time out. Check your SIP server for a reasonable value. Outbound Proxy Outbound proxy is a device that receives requests from a client, even though it may not be the server resolved by the Request-URI. Outbound proxy will forward SIP requests and frequently RTP media traffic to another SIP server. Check with your SIP server (SIP provider) if an outbound proxy is required. Home Domain SIP Networks sometimes use the Home Domain name as an identifier. Enter this field if it is required. Authentication ID Enter the Authentication ID as provided. Password Enter the authentication password as provided. Call Forward Type Call forward can be set under different conditions: Unconditional Forward, Busy Forward, No Answer Forward, Busy or No Answer Forward. Select the call forward type and enter the phone number that you would like the call to be forwarded to. Call Forward Number Enter the phone number that you would like the call to be forwarded to when Call Forward is set. Backup Server ) D E ) F) G) ) H I) ) J K) L) 19 One-Channel GSM VoIP Gateway The GoIP1 Gateway supports one Backup Server as an alternative to the main server. When the registration to the main server fails, the GoIP1 Gateway will try to register to the Backup Server. 3.4.2 Advanced Settings Click on “Advance Settings” tab on the top right corner of the Call Setting page to display all the parameters for programming, as shown below. These parameters allow more advanced control over the SIP signaling and media preference. )Local Signaling Port (SIP Local port) The default SIP port is 5060. Change this as required. B)SIP 183 A Check the box of SIP 183 if the SIP server supports this feature. C) NAT Keep-alive 20 One-Channel GSM VoIP Gateway The NAT Keep-alive feature sends a null packet to the SIP Proxy periodically in order to keep the NAT open on your firewall for incoming data traffic. 3.4.3 Advanced Timing A) No Answer Expiry(32-180s), NICT Expiry(2-180s), Retransmit T1(200-2000ms), Retransmit T2(2000-8000ms) Some SIP proxies may have special timing requirements. Change these parameters as required. B DTMF Signaling 1 DTMF TYPE DTMF signals can be sent over to the called party after a call is established. GoIP1 Gateway supports both Inband and Outband DTMF signal types. ) ) For Inband DTMF type, DTMF signals are generated locally at the calling phone and then 21 One-Channel GSM VoIP Gateway send to the called party as part of the voice signals. This method is not reliable since the quality of the DTMF signals is subject to the Codec used and the quality of the networkt. For Outband DTMF type, DTMF signals are independently translated and sent to the called party. After receiving DTMF signals, the called party translates and interpret based on the DTMF protocol. This method allows more reliable DTMF signaling. However, it requires the called party to support this feature in order for this to work properly. GoIP1 Gateway supports both RFC2833 and SIP INFO DTMF protocols. 2 DTMF Payload Type DTMF Payload Type is defined by RFC2833 protocol to carry the tone definitions for various applications. The default DTMF Payload Type is 101. Please consult your VoIP service provider for the proper setting if required. C Signaling Qos ) ) Signaling QoS improves the performance of SIP signaling. If local network device supports Qos, select this field accordingly. Please consult your network administrator for further information. D Signaling Encryption GoIP1 Gateway supports different encryptions for SIP signaling. Select the one that you prefer. Depending on your network environment and SIP Server capabilities, this feature may or may not be turned on. ) )None Select None to turn off this feature. 2)STUN (RFC 3489) 1 STUN (Simple Traversal of UDP (User Datagram Protocol) through NATs (Network Address Translators)) is a network protocol allowing a client behind a NAT (or multiple NATs) to find out its public address, the type of NAT it is behind and the internet-side port associated by the NAT with a particular local port. Select STUN (RFC 3489) to use a STUN server for Signaling NAT Traversal. Enter the IP Address or the domain name of the STUN server to be used. 2) Relay Proxy 22 One-Channel GSM VoIP Gateway Relay proxy is a proprietary NAT traversal technology. Please consult your service provider for more information. 3.4.4 Media Setting Click on “Media Settings” in the “Call Setting” menu to access the parameters available for media settings. )RTP Port Range A This parameter specifies the range of the RTP (Real Time Protocol) Ports used by the GoIP1 Gateway. If your network limits the usable port range, this parameter may need to be modified. Please consult your network administrator for more information. B Packet Length(ms) This parameter defines the voice packet length. The default setting is 20ms. The range is from 5ms to 40ms at an increment of 5ms. Please note that some codes have a minimum packet length of more than 5ms. C Jitter Buffer Mode ) ) Since data packets may arrive at different orders, the Jitter Buffer is used to hold the data packets received for re-arrangement according to the packet sequence number. Three Jitter Buffer Modes are supported: Adaptive, Sequential, and Fixed. The default is set to Fixed mode with the fixed delay of 60ms. Please consult your network administrator for more information on the network environment in order to determine the optimal settings. 23 One-Channel GSM VoIP Gateway ) D Media Qos Similar to the Signaling QoS, the Media Qos in intended to improve the voice performance or quality if your local network supports QoS. E Media Encryption GoIP1 Gateway supports different encryptions for voice media. F Symmetric RTP Normally GoIP1 Gateway uses RTP ports based on the configuration. If this box is checked, GoIP1 Gateway will identify RTP ports from the media traffic it has received and use the same ports when sending media traffic. G) Media NAT Traversal Similar to Signaling NAT Traversal, this feature allows media packets (RTP) to be routed properly in various network environments. ) ) )None Select None to disable this feature. 2)STUN (RFC 3489 ) 1 STUN (Simple Traversal of UDP (User Datagram Protocol) through NATs (Network Address Translators)) is a network protocol allowing a client behind a NAT (or multiple NATs) to find out its public address, the type of NAT it is behind and the internet-side port associated by the NAT with a particular local port. Select STUN(RFC 3489) to use a STUN server for Signaling NAT Traversal. Enter the IP Address or the domain name of the STUN server to be used. Port forwarding Support Port forwarding (sometimes referred to as tunneling) is the act of forwarding a network port from one network node to another. This technique can allow an external user to reach a port on a private IP address (inside a LAN) from the outside via a NAT-enabled router. In order for this feature to work, the local network Gateway must support this feature and be set up properly. Please consult your network administrator for help to enable this Port forwarding feature. ) 3 24 )Relay Proxy One-Channel GSM VoIP Gateway 4 Relay proxy is a proprietary NAT traversal technology. Please consult your service provider for more information. Codec Preference allows a user to select the codes to be used and its priority for a voice call. 25 One-Channel GSM VoIP Gateway Click on the check box to enable a codec. Select a codec and then press the UP or DOWN button to move the position of the codec on the codec list with a priority in descending order. Note: The voice code alaw and ulaw is G.711a and G.711u. 3.5 Call Divert The Call divert feature controls the routing of calls between VoIP and GSM. 3.5.1 Call Forward (From VoIP To PSTN) 26 One-Channel GSM VoIP Gateway Forward Number Enter a phone number in this field will forward all incoming VoIP calls to this phone number (PSTN or Mobile). Using “,” to add a 500ms delay to the dialing sequence. When Forward Number field has a phone number, GoIP1 will automatically forward all VoIP calls to this phone number. When Forward Number is empty, GoIP1 will route phone calls based on the following conditions. A: When the Callee ID is GoIP1’s SIP account number, GoIP1 will take the call and feed back a dial tone to VoIP caller. Then VoIP caller must dial a PSTN number when hearing this dial tone. B: When the Callee ID is not GoIP1’s SIP account number, GoIP1 will automatically dial out with this number thru GSM network, based on the rules in Dial Plan(VoIP to PSTN) field. Dial Plan Please refer to 3.9 Dial Plan for details. If “:” is entered in the field, all of the phone calls will pass through. Forward to PSTN Auth Mode This field sets protection for using GoIP1 to connect to GSM network. 1) No Auth Anyone can make phone calls through GoIP1. 2) Password If a password is entered, the GoIP1 will generate an indication tone and wait for the caller to dial the password. 3) Trust List 27 One-Channel GSM VoIP Gateway 4) Enter the phone numbers on the Trust Number field if Trust List is used. People calling from the trust phone numbers will be able to use GSM connection. Password or Trust List Callers will be able to use GoIP1 for GSM connection if their phone numbers are on trust phone number list or if they have the password. SIM Card Settings 1) 2) 3) 4) SIM Card Expiry - usage limit (minutes) SIM Card State Report Number - the recipient phone number for the SMS report SIM Card State Report Time - the time schedule to send SMS report SIM Card ID - Identification sent with the sms message 3.5.2 Call Forward (From PSTN To VoIP) Forward Number 28 One-Channel GSM VoIP Gateway Enter a phone number in this field will forward all incoming PSTN (GSM) calls to this phone number (a VoIP number). If this field is blank, the GoIP1 answers all incoming GSM calls and then generates the dial tone. The caller can then dial a VoIP number. When Dial Plan Please refer to 3.9 Dial Plan for details. Forward to VoIP Auth Mode This field sets protection for using the GSM connection to VoIP. 5) No Auth Anyone can make phone calls through GoIP1. 6) Password If a password is entered, the GoIP1 will generate an indication tone and wait for the caller to dial the password. 7) Trust List 8) Enter the phone numbers in the Trust Number field if Trust List is used. People calling from the trust phone numbers will be able to use GoIP1 to connect to VoIP. Password or Trust List Callers will be able to use GoIP1 for VoIP connection if their phone numbers are on trust phone number list or if they have the password. 29 One-Channel GSM VoIP Gateway 3.6 SMS Disposal 3.6.1 SMS Call Out GoIP1 Gateway supported SMS call. In this mode, when GoIP1 Gateway receives a SMS from a mobile phone, it will automatically make a call to SIP server. To use this function, select the SMS Dial option in configuration page. GoIP1 supported three types SMS Dial::: A Mode 1 GoIP1 dial the call use SMS sender call ID B Mode 2 GoIP1 dial the call via its VoIP account and add the SMS sender phone number to Call Divert option’s Forward Number (VoIP to PSTN) automatically. C Mode 3 GoIP1 dial the call via its VoIP account and add the SMS sender phone number to SIP invites header.