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MasterVoIP

VoIP to GSM Gateway

3.4.5.2 SIM Settings

After making the port settings, press a

SIM

icon in the left pane. The

SIM

Setting window opens

VoIP Master

. In the PIN Code box, enter the PIN number of the SIM.

2. In the Network box, enter the GSM network number of the SIM.

3. In boxes 1 through 8, set enter telephone number prefixes to which this SIM can dial.

4. Press

Write Settings

Содержание GSM gateway

Страница 1: ...VoIPMaster Version 4 x VoIP to GSM gateway Connecting Cellular Phones directly to Voice over IP worldwide networks User Manual ...

Страница 2: ...dance with all applicable codes regulations and safety measures Trademark and Patents All trademarks patents and copyrights apply General Manual Notes Without notice and without obligation the contents of this manual may be revised to incorporate changes and improvements Every effort has been made to ensure that the information in this manual is most complete and accurate while writing this time o...

Страница 3: ...value and reliability of products Warranty Policy The Dual Cell to BRI Gateway product you have purchased is under warranty of 12 months from the date of purchase by the original purchaser In case of defects of materials or workmanship Eurotech Communication will replace it free of charge This warranty applies to hardware software but does not include SIM Cards This warranty will not be honoured i...

Страница 4: ...3 1 7 LED Light Pattern Indication 14 3 2 Configuration Guide 15 3 2 1 Configuring VOIP Client with a Web Browser 15 3 2 1 1 Access the Web Configuration Menu 15 3 2 1 2 End User Configuration 15 3 2 1 3 Advanced User Configuration 19 3 2 1 4 Saving the Configuration Changes 29 3 2 1 5 Remotely rebooting VoIP Client ATA 29 3 3 Restoring the Factory Default Settings 30 3 4 VoIP Master 31 3 4 1 What...

Страница 5: ...guration interface The VoIP Client ATA provides VoIP call origination and termination with PSTN network with some add on supplementary services which are reviewed at Chapter 3 The VoIPMaster gateway adds new capabilities of GSM to VoIP calls origination and termination to the client ATA The following topics of the VoIP Client ATA are reviewed in Chapter 3 Client ATA Product Overview Key Features H...

Страница 6: ...ettings for the GSM Port to define policies and profile of behaviour when dialling SIM Settings regarding with usage limits and other optional modes Call follow me settings to let the system call you while you are away from office as if you where in office Call Back Settings to let waiting lines make the call when line is available again At menu Cellular Gateways Please Give Us feedback to improve...

Страница 7: ...Hardware Device The VoIP Master Gateway 110 220V Electric Power converter to 24V with cables supplied VoIP master software Installation CD Installation kit for MS Windows Management Application this User Manual file and additional auxiliary utilities GSM Antenna To be installed to the VoIP Master Gateway RS 232 Serial PC COMport connection cable we ll be referred as Comport cable in this manual ...

Страница 8: ...emo provided equipment regarding with standards support details Supports SIP 2 0 RFC 3261 TCP UDP IP RTP RTCP HTTP ICMP ARP RARP DNS DHCP both client and server NTP PPPoE STUN TFTP etc Powerful digital signal processing DSP to ensure superb audio quality advanced adaptive jitter control and packet loss concealment technology Supports various codecs including G 711 PCM a law and u law G 723 1 5 3K ...

Страница 9: ...able describes the hardware specification of VoIP Client ATA Model VoIP Client ATA LAN interface 1xRJ45 10Base T Button 1 LED GREEN RED color Universal Power Adaptor Input 100 240VAC Output 5VDC 1200mA UL certified Dimension 65mm W 93mm D 27mm H Weight Operating Temperature 32 104oF 0 40oC Humidity 10 95 non condensing Compliance FCC CE C Tick ...

Страница 10: ...option 02 04 Gateway IP address Same as Menu option 02 05 DNS Server IP address Same as Menu option 02 06 TFTP Server IP address Same as Menu option 02 TFTP server is used to update the firmware of the device 47 Direct IP Calling When entered user will be prompted by dial tone dial the 12 digit IP address to make a direct IP call For details see 4 2 2 Make a Direct IP Call 86 No Voice Messages or ...

Страница 11: ... ATA and the other VoIP device i e another VOIP Client ATA or other SIP products can be connected through a router using public or private IP addresses To make a direct IP call first pick up the analog phone or turn on the speakerphone on the analog phone then access the voice menu prompt by dial or press the button on the HT286 and dial 47 to access the direct IP call menu User will hear a voice ...

Страница 12: ...y to recover the call The busy tone is just to indicate to the transferor that the transfer has failed Busy tone keeps playing This means we have failed to receive the second NOTIFY from the transferee and decided to time out Note this does not indicate the transfer has been successful nor does it indicate the transfer has failed When transferee is a client that does not support the second NOTIFY ...

Страница 13: ...n hang up 91 Cancel Busy Call Forward To cancel Busy Call Forward dial 91 and get the dial tone then hang up 92 Delayed Call Forward To use this feature dial 92 and get the dial tone Then dial the forward number and for a dial tone then hang up 93 Cancel Delayed Call Forward To cancel this Forward dial 93 and get the dial tone then hang up Flash Hook When in conversation this action will switch to...

Страница 14: ...cates abnormal status DHCP Failed or WAN No Cable flash every 2 seconds if DHCP is configured VOIP Client 486 fails to register flash every 2 seconds if SIP is configured GREEN LED indicates normal working status Message Waiting Indication Button flashes every 2 seconds RINGING Button flashes at 1 10 second RINGING INTERVAL Button flashes every second ...

Страница 15: ...hrough section 2 1 with menu option 02 Then access the VOIP Client s Web Configuration Menu using the following URI http Phone IP Address where the Phone IP Address is the IP address of the phone 3 2 1 2 End User Configuration Once this request is entered and sent from a Web browser the IP phone will respond with the following login screen The password is case sensitive with a maximum length of 25...

Страница 16: ... The HT 286 will attempt to establish a PPPoE session if any of the PPPoE fields is set In this mode the WAN side web access is disabled and TFTP upgrade for firmware is not feasible and HTTP upgrade is the only available solution If Static IP mode is selected then the IP address Subnet Mask Default Router IP address DNS Server 1 primary DNS Server 2 secondary fields will need to be configured The...

Страница 17: ...Februaray 2006 Page 17 42 MasterVoIP VoIP to GSM Gateway Here are the status details shown ...

Страница 18: ...his is the user interface normally not changed VOC This is the codec program normally not changed System Uptime This shows system up time since last reboot Registered This shows whether the unit is registered to service provider s server PPPoE Link Up This shows whether the PPPoE is up if connected to DSL modem NAT This shows what kind NAT the VoIP Client ATA is connected to via its WAN port It is...

Страница 19: ...lowing page The password is case sensitive with a maximum length of 25 characters and the factory default password for Advanced User is admin Advanced User configuration page includes not only the end user configuration but also some advanced settings such as SIP configuration Codec selection NAT Traversal Setting and other miscellaneous settings Following is the screen shot of the Advanced config...

Страница 20: ...Februaray 2006 Page 20 42 MasterVoIP VoIP to GSM Gateway The following window if for advanced configuration regarding IP SIP QoS NAT IP Telephony modes setting ...

Страница 21: ...SIP Server This field contains the URI string or the IP address and port if different from 5060 of the SIP proxy server e g the following are some valid examples sip my voip provider com or sip my company sip server com or 192 168 1 200 5066 Outbound Proxy This field contains the URI string or the IP address and port if different from 5060 of the outbound proxy If there is no outbound proxy this f...

Страница 22: ...Februaray 2006 Page 22 42 MasterVoIP VoIP to GSM Gateway ...

Страница 23: ...d Vocoder VoIP Client ATA supports up to 7 different vocoder types including G711 ulaw PCMU G711 alaw PCMA G723 G729A B G726 32 ADPCM G728 and iLBC Depending on the product model some of these vocoders may not be provided in a standard release A user can configure vocoders in a preference list that will be included with the same preference order in SDP message The first vocoder in this list can be...

Страница 24: ...se codec PCMU PCMA User ID is phone number If the VoIP Client ATA has an assigned PSTN telephone number then this field will be set to Yes Otherwise set it to No If Yes a user phone parameter will be attached to the From header in SIP request SIP Registration This parameter controls whether the IP phone needs to send REGISTER messages to the proxy server The default setting is Yes Unregister On Re...

Страница 25: ...SIP and RTP ports This is usually necessary when multiple IP phones are behind the same NAT keep alive interval The VoIP Client ATA sends a UDP package to the SIP server periodically in order to keep the port open on the router This parameter defines the interval time that HT286 send the UDP package The default setting is 20 second Use NAT IP NAT IP address used in SIP SDP message Default is blank...

Страница 26: ...t to No VoIP Client ATA will only do HTTP download once at boot up Automatic HTTP Upgrade Choose Yes to enable automatic HTTP upgrade and provisioning In Check for new firmware every field Enter the number of days period VoIP Client ATA will check the HTTP server for firmware upgrade or configuration after the defined number of days When set to No VoIP Client ATA will only do HTTP upgrade once at ...

Страница 27: ... Denmark Onhook Voltage Select the onhook voltage to suit different area or PBX Polarity Reversal Select Polarity Reversal to adapt some call charge billing system Default is No NTP server This parameter defines the URI or IP address of the NTP server which the IP phone will use to display the current date time Send Anonymous If this parameter is set to Yes the From header in the outgoing INVITE m...

Страница 28: ...d SIP message DEBUG level SIP message summary INFO level inbound and outbound calls INFO level registration status change INFO level negotiated codec INFO level Ethernet link up INFO level SLIC chip exception WARNING and ERROR levels memory exception ERROR level The Syslog uses USER facility In addition to standard Syslog payload it contains the following components GS_LOG device MAC address error...

Страница 29: ...Users are recommended to power cycle the VOIP Client 488 after seeing the above message 3 2 1 5 Remotely rebooting VoIP Client ATA The administrator of the phone can remotely reboot the phone by pressing the Reboot button at the Configurations menu button Once done the following screen will be displayed to indicate that rebooting is underway At this point the user can relogin to the phone after wa...

Страница 30: ...ess of the device The MAC address of the device is located at the bottom of the device It is a 12 digits hex number Step 2 Encode the MAC address to decimal digits Please use the following mapping 0 9 0 9 A 22 B 222 C 2222 D 33 E 333 F 3333 For example for the MAC address 00 0b 82 00 e3 95 the User encoding should be 00 0222 82 00 333 3 95 Step 3 Access the voice menu by pressing or the LED button...

Страница 31: ...dule including a SIM card is installed inside the VoIP device A SIM card is a smart card that is received with a subscription to a cellular telephone network This following is the communication solution architecture enabled by the VoIPMaster One Location in the World Another Location in the World IP Network VoIP Near 0 Cost GSM Base Stations GSM Cellphones GSM Cellphones VoIPMaster Management VoIP...

Страница 32: ... to 32 cellular phones can use the VoIP device for connection to the internet in parallel A local desktop telephone can be connected to the VoIP device The desktop phone can send and receive calls via the internet as well as via the GSM network according to telephone prefixes A follow me function can be activated to serve the desktop phone If two systems install in remote offices a call from a mob...

Страница 33: ...de of a Free Gate 2 Connect cables as follows a Insert the antenna to a connector on the right side of the VoIP Master b Insert the communication cable from the PC COMport to the serial COM port socket on the left side of the VoIPMaster c Insert the network cable into a socket on the right side of the gateway and connect it to the computer with the internet connection d Insert the telephone jack f...

Страница 34: ...Februaray 2006 Page 34 42 MasterVoIP VoIP to GSM Gateway ...

Страница 35: ...ter gateway Configuration is done by a manger application in the computer Install the manager application on the software cd then define the comport connection as described in this chapter 1 Insert the VoIP Master CD into the computer drive 2 In Windows Explorer navigate to Icon in the software disk 3 Double click the Icon wait till the installation window will open ...

Страница 36: ...nstallation window opens VoIP Master 6 Click Install Wait till the VoIP Master Manager application will install itself 3 4 4 Define the Com port Connection After installing the manager application launch it and define the Comport to which the VoIP Master is connected 1 Launch the PRI Manager by pressing on your computer desktop or by pressing ...

Страница 37: ... VoIP Master 2 In the toolbar press The Select Connection window opens 3 Select the Com Port in the computer to which the VoIP Master is connected The connection indicators in the lower right corner of the window blink green After installing the Manager and defining the port connection define port and SIM Settings as described in the following chapter ...

Страница 38: ...Februaray 2006 Page 38 42 MasterVoIP VoIP to GSM Gateway ...

Страница 39: ... settings in this window as follows 1 In the Dial pause box set the time interval whereupon a dialed number is dispatched after the designated delay time Each unit is 0 1 second For example if you want the number to be dispatched 3 seconds after you finish dialing enter 30 in this box 2 Upon completion of a call if you want to remain connected to the GSM Network set Repeat Access to VoIP to On 3 S...

Страница 40: ...tings press a SIM icon in the left pane The SIM Setting window opens VoIP Master 1 In the PIN Code box enter the PIN number of the SIM 2 In the Network box enter the GSM network number of the SIM 3 In boxes 1 through 8 set enter telephone number prefixes to which this SIM can dial 4 Press Write Settings ...

Страница 41: ...all to your cellular phone To activate after making SIM settings press Follow me in the left pane The Follow Me Setting window opens VoIP Master 1 Set the Mode box to ON 2 In the Rings Number box enter the number of times the local phone will ring before being diverted to the Follow me function 3 In the Called Number box enter the telephone number that you want dialed when the follow me function i...

Страница 42: ... person B is listed in the Call Back settings of the VoIPMaster manager when the phone call of person A is completed the VoIPMaster gateway will call person B and provide a telephone line that was previously busy by person A To enable this feature perform the following 1 On the right side of the window set the box to ON 2 Enter desired telephone numbers in the center of the window 3 Next to each t...

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