GSM VOIP Gateway Series User Manual
GSM VOIP Gateway Series 1. OVERVIEW 1.1 Introduction GSM VOIP Gateway Series Color VoIP Channel GSM Channel Gray 1 1 Gray 4 4 Hardware Parameter Remark Model WT-2208 Customized The GSM VOIP Gateway is a broadband relay gateway newly developed by Processor ARM9E 133Mhz VinTelecom Technology. It is a new product for seamless connection between DSP VPDSP101 95Mhz the GSM network and VoIP network.
GSM VOIP Gateway Series GSM VOIP Gateway Series 1.2 Select Tool > Reset to restart the GSM VOIP Gateway. 4 Parameters Of Equipment Hardware Feature Parameter Parameter Remark Model Processor DSP RAM FLASH Power WT-2201 ARM9E 133Mhz VPDSP101 95Mhz 16M 4M DC12V/2A ±10% WT-2204 ARM9E 133Mhz VPDSP101 95Mhz 16M 4M DC12V/2A ±10% WT-2208 GSM Band Default 900M/1800M Optional 850/1900M Max. 5W RUN, GSM, LAN, PC 2 0.10KG 0-40°C Default 900M/1800M Optional 850M/1900M Max. 12W RUN, GSM, LAN, PC 2 0.
GSM VOIP Gateway Series GSM VOIP Gateway Series Note: During the upgrading, do not cut off the power. Otherwise, the GSM VOIP 1.4 Software Feature Gateway will be damaged. LINUX OS Embedded HTTP that accesses internal parameters PPPoE diali ng You can modify the password of the user and administrator. Select Tool > Modify NAT broadband routing function Password. The password modification page is displayed, as shown in the following DHCP cli ent figure.
GSM VOIP Gateway Series GSM VOIP Gateway Series Note: Some of parameters of the gateway wil l not be valid until the gateway is restarted. Therefore, you are advised to restart the gateway after the parameters are modified, so that the modification can take effect. 3.12 Abandon the change When the new setting is not saved, you can clear all the unsaved parameters. 3.13 Tool Select Menu > Tool. The following page is displayed.
GSM VOIP Gateway Series 3. GSM VOIP Gateway Series PC port supports network sharing, so connect the PC port to the computer or lower-layer switch (HUB or router) 4. 2.2 Connect the output terminal of the transformer with the power port. Connection Figure 3.11 Save the Change After setting is changed, click “Save” and the new setting will be valid. Otherwise, the new setting is invalid. GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: technical@witura.
GSM VOIP Gateway Series GSM VOIP Gateway Series SIP Terminal Disable: It’s not all owed to transfer the PSTN caller number to the VoIP system; Enable: The CID is set as the SIP caller number. Use Remote Party ID: The GSM VOIP Gateway will add the PSTN caller number to the call request signaling of the VoIP system. The signaling is as follows (provided that the PSTN caller number is 13800000000): 2.
GSM VOIP Gateway Series GSM VOIP Gateway Series To: Call-ID: 808807EB-A8B3-DD11-BBA6-005056C00008@192.168.2.89 CSeq: 3MESSAGE Contact:
GSM VOIP Gateway Series GSM VOIP Gateway Series The following is an example that the GSM VOIP Gateway forwards the SMS to the Users can send instructions to the GSM VOIP Gateway through the SMS. SIP 3999. The red part is the content of the SMS. Function MESSAGE sip:3999@192.168.2.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.162:5060;branch=z9hG4bK1967685528 From: ;tag=667435795 Obtain information from the LAN port Reset the GSM VOIP Gateway setting To:
GSM VOIP Gateway Series 3.1 GSM VOIP Gateway Series Page Setting Menu canMapAlias = FALSE You can access the setting page of the GSM VOIP Gateway through the IP callIdentifier = { address of the LAN port or PC port. The default factory settings are as follows: guid = 16 octets { A: The LAN port supports the DHCP (dynamic IP address). Users can dial the SIM card number of the gateway and if connected dial *00 to obtain the IP cb 40 a4 af 8e 9b 60 96 6b 5f a0 03 f2 ed 55 5b .@....`.k_....U[ address.
GSM VOIP Gateway Series GSM VOIP Gateway Series [0] = dialedDigits "9998675588228822" } srcInfo = 2 elements { [0] = dialedDigits "20001" [1] = h323-ID "20001" } srcCallSignalAddress = ipAddress { Click “OK” button and the gateway status page is displayed as default ip = 4 octets { c0 a8 02 ed .... } port = 2049 } bandWidth = 2048 WT-2201 Status Interface callReferenceValue = 7502 conferenceID = 16 octets { 7f f3 78 77 49 3f 4c c1 9a dc 6a 84 12 d8 30 8f ..xwI?L...j...0.
GSM VOIP Gateway Series GSM VOIP Gateway Series activeMC = FALSE answerCall = FALSE canMapAlias = FALSE callIdentifier = { guid = 16 octets { cb 40 a4 af 8e 9b 60 96 6b 5f a0 03 f2 ed 55 5b .@....`.k_....
GSM VOIP Gateway Series callType = pointToPoint NULL GSM VOIP Gateway Series You can also access the setting page of the GSM VOIP Gateway through the IP address 192.168.2.216 or 192.168.2.172 of the LAN port of the gateway. The endpointIdentifier = "3705_endp" login method is the same as that of the PC port, but you must first obtain the IP address of the LAN port. destinationInfo = 1 elements { [0] = dialedDigits "8675588228822" 3.
GSM VOIP Gateway Series GSM VOIP Gateway Series A. Examples of SMS Dialing: LAN Port In the following SMS dialing examples, the H.323 number of the GSM VOIP It displays the current IP address of the LAN port, such as 192.168.2.172. B. Gateway is set as follows: PC Port It displays the current IP address of the PC port. C. PPPoE Dialing It displays the PPPoE broadband connection condit ion. After the connection, the IP address obtained is displayed on the LAN port. D.
GSM VOIP Gateway Series GSM VOIP Gateway Series Content-Length: 226 3.9.2 SMS Dialing under the H.323 Protocol The GSM VOIP Gateway permits users to dial back through the SMS under the H.323 protocol. After users send the called number to the GSM VOIP Gateway through the SMS, the GSM VOIP Gateway will send a call request to the H.323 GK automatically. Users who need this function shall choose the following parameters: User Options of the WT-2204/ WT-2208 3.3.
GSM VOIP Gateway Series 3.3.2 Time Zone and Time Server GSM VOIP Gateway Series GSM VOIP Gateway sends a call request through the SIP number of the GSM VOIP Gateway, the GSM VOIP Gateway will automatically add the number of the short message sender to the PSTN Forwarding Number in Cal Forwarding (VoIP This item displays the adjusted time according to the selected time zone.
GSM VOIP Gateway Series GSM VOIP Gateway Series SendingMessage to 192.168.2.1:5060: INVITE sip:8675588228822@192.168.2.1:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.2.237:5060;branch=z9hG4bK363969813 It is a special server, which needs the support of the specific system. From: ;user=phone;tag=65248630 To: Call-ID: 117025903@192.168.2.237 3.3.
GSM VOIP Gateway Series GSM VOIP Gateway Series Users can establish a GSM group containing multiple GSM VOIP Gateway. Under this mode, the administrator only needs to provide a GSM number to the user to call in the VoIP system. From: ;user=phone;tag=65248630 To: Call-ID: 117025903@192.168.2.237 CSeq: 2 INVITE Contact:
GSM VOIP Gateway Series GSM VOIP Gateway Series Sending Message to 192.168.2.1:5060: INVITE sip:8675588228822@192.168.2.1:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.2.237:5060;branch=z9hG4bK363969813 Serve as the client: When GSM VOIP Gateway operates in this mode, it will send it s real-time status to the server of the GSM VOIP Gateway group, so that From: the GSM VOIP Gateway server can deploy the call forwarding.
GSM VOIP Gateway Series GSM VOIP Gateway Series C: Mode 3 In this mode, the GSM VOIP Gateway sets the SIP number of the GSM VOIP Gateway as the calling number of the call and the called number as the short message content and the number of the short message sender, whose format is short message content*the number of the short message sender.
GSM VOIP Gateway Series GSM VOIP Gateway Series 3.9 3.3.10 Timing Restart SMS Mode The GSM VOIP Gateway restarts at least once at the specified time every day to The GSM VOIP Gateway permits you to call VoIP users or forward short messages clear the buffer of the GSM VOIP Gateway, so that the GSM VOIP Gateway can through the SMS. operate normally. 3.9.1 SMS Dialing Under SIP Protocol Under the SIP protocol, the GSM VOIP Gateway permit s users to dial back through the SMS.
GSM VOIP Gateway Series GSM VOIP Gateway Series SIM card status reporting number: The gateway can report the status of SIM cards (remaining call duration) through the SMS. This parameter is used to specify the mobile phone number to receive the SMS. SIM card status reporting time: This parameter is used to specify the remaining call duration and then send the report. SIM card ID: This parameter is used to specify the ID of SIM cards in the short message report.
GSM VOIP Gateway Series GSM VOIP Gateway Series This mode is used to set the above password authentication and trust list authentication at the same time. c3on: the duration when the frequency 3 is on (ms) For a downlink call, the authentication mode is as follows: If the number in the C3off: the duration when the frequency 3 is off (ms) trust list is used to dial the user served by the PSTN, the call will be connected.
GSM VOIP Gateway Series GSM VOIP Gateway Series Downlink 3.4.1 LAN Port Setting Uplink The setting is as follows: Select “Forward to PSTN Authentication Mode” > “Trust List Authentication”. Click the “VoIP Trust Number List”, and the VoIP Trust The LAN port of the GSM VOIP Gateway can be set to the dynamic IP through Number List is displayed (Maximum 15 trust numbers can be entered). Enter the DHCP, fixed IP, and PPPoE dialing.
GSM VOIP Gateway Series GSM VOIP Gateway Series Set these parameters according to the network the user uses. 3.7.2 Authentication Mode Setting C. PPPOE The authentication mode is classified into the password authentication, trust list PPPoE (Point-to-point protocol over Ethernet) is a network protocol that authentication, and password or trust list authentication. compresses the PPP in the Ethernet. Select PPPoE dialing, and enter the account and password provided by the network provider.
GSM VOIP Gateway Series GSM VOIP Gateway Series B. Fixed IP Address 3.7 Select the fixed IP, and the following setting parameters are dis played. Enter the Call Forwarding (Setting Authentication Mode) on the Call Route And IP address and subnet mask (the network section of the IP address should be different from that of the LAN port to prevent conflict). The gateway provides the call routing function for users, which can be set in the Call Forwarding Setting.
GSM VOIP Gateway Series GSM VOIP Gateway Series The rule is “0:|13[0-9]xxxxxxxx:+0|[1-8]xxxxxxx:+0755”. When the main DNS address fail s to connect or is not avail able, the secondary DNS can be used (such as 202.67.156.222 or obtain from the service provider). If When you dial the number 88990011 and 8899001133, the result is the same. the PPPoE is set, the secondary DNS will be automatically provided by the service The number actually dialed is 075588990011. provider. This parameter can be null .
GSM VOIP Gateway Series GSM VOIP Gateway Series B. Display Name This parameter is used to display the name of the user who subscribes the H.323 3.5.7.2 Dialing Rule with Specified Length Of Numbers service. For example, when you call your friend John Smith, your name will be displayed on your friend’s telephone. If you need to specify the length of telephone numbers matched, you can specify the dialing rule as “AAXXXXXX:-aa+bb”. Where, “AAXXXXXX” indicates the C. H.
GSM VOIP Gateway Series GSM VOIP Gateway Series added. If the number fails to match, the number continues to match the next Under the Gatekeeper mode, the GSM VOIP Gateway operates in the H.323 rule. If no digit after the colon is specified, such as “00:”, it indicates that no register status. When you register through the H.323 protocol, select “H.323 actions are taken when “00” is matched and the number exit s the matching. Terminal” in “Terminal Type”, as shown in the above figure.
GSM VOIP Gateway Series GSM VOIP Gateway Series Under the Gatekeeper mode, the GSM VOIP Gateway will forward all calls to the C. STUN (RFC 3489) VoIP network to this address. The Simple Traversal of UDP over NAT (STUN) is a protocol that enables the SIP telephone to detect the existence and type of the firewall installed in the Note: The value of this parameter must be standard ASCII characters. computer. This parameter indicates the SIP address of the STUN server.
GSM VOIP Gateway Series GSM VOIP Gateway Series C. Trunk Agent The trunk agent protocol is a firewall traversal technology developed by Witura Corporation Sdn Bhd. It enables the products of Witura Corporation Sdn Bhd to be applicable for most LANs. It involves the address, port, user name and password. A. RAS Port The RAS is the communication protocol between the terminal and the Gatekeeper.
GSM VOIP Gateway Series GSM VOIP Gateway Series F. H245 Tunnel This parameter is set for the special requirements of some customers. If you are not sure, do not set this parameter. G. Registration Mode This parameter is used to comply with different PBXs and is not set normally. H. DTMF Signals DTMF signals are used to transmit call signals to the call switching center over the audio band. The DTMF means that two different frequencies of sounds are combined to 16 types of dialing tones.
GSM VOIP Gateway Series GSM VOIP Gateway Series In the advance option of the call setting, the signaling and media have separate firewall setting, as shown in the following figures. 3.5.6.1 Traversal of H323 Signaling Over NAT The traversal of H323 signaling over NAT (firewall) is classified into 4 categories: 3.5.2 SIP Phone The SIP (Session Initiation Protocol) is a simple network protocol that has less hierarchy and facilitates the initiation of calls among users.
GSM VOIP Gateway Series 3.5.2.2 GSM VOIP Gateway Series Single Server Mode C. Jitter Delay Processing Mode This parameter is used to specify the algorithm model of the jitter delay buffer. The “Adaptive” mode should be set. Other modes are only used for tests and should not be set in actual applications. D.
GSM VOIP Gateway Series GSM VOIP Gateway Series 2) Out-band DTMF E) Homing Domain The out-band DTMF transmit s dialing tones over protocols, such as RFC2833 and This parameter is used for the domain management host of the SIP (a host that SIP INFO, which can ensure the validity of the transmission. provides the SIP service). F.
GSM VOIP Gateway Series 3.5.2.3 Setting by Line (Valid for the WT-2204 and WT-2208) GSM VOIP Gateway Series C. Timeout Setting D. Signaling QoS Quality of Service (QoS) is a network’s capacity to provide priority services, including the special bandwidth, jitter control and delay (used for real-time and interactive traffic), and improvement of the packet loss ratio. This parameter is used to mark the specified QoS label for the call signaling packet to increase the network service quality.
GSM VOIP Gateway Series GSM VOIP Gateway Series B) Gateway Prefix The gateway prefix enables the connection of call s through a particular line. It 3.5.3 SIP Advance Setting can match the first digit only. You can set a gateway prefix for multiple lines. When you set a gateway prefix for multiple lines, the call s that have the same The advance setting of the SIP involves the signaling and media. Users can set gateway prefix will select the line set with this gateway prefix.
GSM VOIP Gateway Series GSM VOIP Gateway Series H) Password A) SIP Trunk Gateway1 This parameter is used to set the authentication password when the gateway logs It is the IP address of the server connected to the GSM VOIP Gateway. When the into the SIP proxy server. registration timeout is 0, the GSM VOIP Gateway is connected to the SIP server.