- Ozeki 10
- Bulk Messenger
- Cluster Software
- Developer tools
- Sierra SMS modem
- Multitech SMS modem
- Wavecom SMS modem
- 8 port GSM modem pool
- VoIP GSM Gateway
- Configure VoIP functionality
- Configure SMS functionality
Introducing the Ozeki VoIP GSM Gateway
The Ozeki VoIP GSM Gateway is a device that lets you make\receive Voice calls and
send\receive SMS messages through the mobile telephone network. This is a 4 channel
device. It means 4 simultanous calls can take place at the same time. This
product has been tested and it works well with Ozeki VoIP products, such as
Ozeki VoIP SIP SDK,
Ozeki NG SMS Gateway or
Ozeki Phone System XE. It can
also be used to notify system administrators through automated Voice Calls when
server breakdowns happen if you use the Ozeki Cluter software to achieve better uptimes.
Please check out the following video to learn more about the Ozeki VoIP GSM Gateway.
|Video tutorial:||How to setup Ozeki VoIP GSM Gateway|
1. What you need
In order to make calls from one of our software through a GSM network you need to connect the Ozeki VoIP GSM Gateway and our software together and you need to insert a SIM card into the Ozeki VoIP GSM Gateway. In the Figure 1 you can see the accessories:
- SIM card: It is used to provide a connection to a GSM service provider. You can buy it directly from a GSM service provider.
- Ozeki VoIP GSM Gateway: It can be used between the GSM network and a few Ozeki software. It is a gateway that allows you to make calls toward the Public Switched Telephone Network from the VoIP phone network. You canorder it directly from Ozeki.
- A communication software: For example Ozeki Dialer. It is a SIP VoIP software. It allows you to make calls toward the GSM network with the help of Ozeki VoIP GSM Gateway. You can order it directly from Ozeki.
Figure 1 - Making calls from Ozeki Bulk Messenger through Ozeki VoIP GSM Gateway
2. Opening the box
After you have received the Ozeki VoIP GSM Gateway package (you can see it in Figure 2) unpack the box.
Figure 2 - Ozeki VoIP GSM Gateway package
The Figure 3 shows the opened box with its content.
Figure 3 - The opened box
Take the accessories out of the box then you check them. The Figure 4 shows what you can find in the box (from right):
- 1× Ozeki VoIP GSM Gateway
- 1× DC Power Adapter
- 1× Ethernet cable
In addition you can see a mini-SIM card (25.00 mm × 15.00 mm × 0.76 mm) that will be inserted into a SIM card slot (Figure 3 on the left).
Figure 4 - The contents of the package + a mini-SIM card
3. Connecting the DC Adapter and a SIM card to each other
After that, you need to connect the Ozeki GSM Gateway to a plug, to the computer network and you need to put one-four SIM card(s) into the device.
The Figure 5 shows the slots where the SIM cards and the DC adapter have to be plugged.
- The DC adapter is plugged into the rightmost connector that is signed with a DC12V label.
- The SIM cards are plugged into the slots which are located between the Channel 1-4 labels and the antennas. The chip on the SIM card has to see up when you put it into the slot.
Figure 5 - DC adapter and SIM card slots 1 (SIM card out)
After you have inserted the SIM cards into the slots you need to push it while a click is not listened (Figure 6).
Figure 6 - DC adapter and SIM card slots 2 (SIM card in)
4. Connecting the Ozeki VoIP GSM Gateway to a computer
If the SIM card and DC adapter have been plugged, you need to connect the Ozeki GSM Gateway to a computer to set the IP address of the device.
To connect a gateway and a computer to each other you need to use a straight-through Ethernet cable (the yellow cable on Figure 7). One of the ends of the cable has to be plugged into the Ethernet port of the computer and the other has to be plugged into the PC port of the Ozeki VoIP GSM Gateway.
Figure 7 - Connecting the Ozeki VoIP GSM Gateway to a PC
Now you can configure the Ozeki VoIP GSM Gateway in a web browser.
By default the IP address of the gateway's PC port is 192.168.8.1. To reach the graphical user interface of the device you need to set an IP address on the Ethernet port of the computer that is in the same network. The IP address of the PC can be from the range of 192.168.8.2 - 192.168.8.254.
5. Network settings in the Ozeki VoIP GSM Gateway
Open a web browser, such as Internet Explorer and enter http://192.168.8.1 in the address bar. Then the login page is popped up (Figure 8) to give the username in the User Name and password in the Password field (root username: admin; root password: admin).
Figure 8 - Login to Ozeki VoIP GSM Gateway
After you have clicked on the Ok button, the Status page will appear. To set the IP address of the gateway, click on the Configurations and Network links. The Figure 9 shows what you need to set on the LAN port:
- LAN port mode: Static IP.
- IP address: An address from the network to which you would like to connect the gateway.
- Subnet mask: The proper mask that belongs to the connected network.
- Default route: The IP address of the default gateway. This is the IP address of that router's interface which is connected to the local computer network.
- Primary DNS: The IP address of the domain name server that is responsible for assigning the IP addresses and the domain names.
Figure 9 - Network settings
Then click on the Save Changes and follow the below instructions:
- Connect the gateway to the local computer network through the LAN port of the device (Figure 10).
- Disconnect the computer from the gateway and connect it back to the local computer network.
- Set the IP address of the computer back.
Figure 10 - Connecting the gateway to the local computer network
After doing these settings, you can configure the Ozeki VoIP GSM Gateway from any computer in the same computer network. Just type the previously set IP address to the web browser, give the login data and you can configure it from anywhere.
6. What do you need to setup in the graphical user interface of the Ozeki VoIP GSM Gateway
If you have inserted the SIM cards into the Ozeki VoIP GSM Gateway, you can see the status of the SIM cards.
In the Figure 11 you can see that there are three SIM cards, which are inserted into the gateway and they are logged in successfully.
Figure 11 - Status of the SIM cards
In Figure 12 you can see the VoIP settings.
From the Config mode menu you need to select the Single Server Mode option. On the right side of this menu, click on the SIP Advanced Settings link. Then choose the None option from the VoIP to PSTN Auth Mode and click on the Save settings button.
Figure 12 - VoIP configuration
- Provide 4 cellular channels for IP-PBX
- Open Standard VoIP Protocols (SIP&H.323)
- Single or Multiple Server Registrations
- Two 10/100 Ethernet for WAN / LAN connections
- Peer-to-Peer IP Calls
- Quad band GSM module: 850/900/1800/1900
- Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
- Line Echo Cancellation
- VLAN and QoS support
- NAT Transversal and Router functions
- Voice prompts, HTTP Web, Auto Provision support for configuration and updates
- Highly stable embedded Linux operating system in high performance ARM 9 Processor
- Processor: ARM9E 133MHz
- DSP: VPDSP101-4 100MHz
- Memory: RAM 16MB/ Flash 4MB
- GSM Module: 850MHz, 900MHz, 1800MHz, 1900MHz
- Power: 12 VDC 2A (110V-220V) (AC/DC adapter included)
- Power consumption: 5W maximum
- Operating temperature: 10°C to 40°C (32°F to 104°F)
- Storage temperature: 0°C to 50°C (32°F to 122°F)
- Size: 195mm (W) x 340mm (L) x 60mm (H)
- Weight: 0.85KG (Including AC/DC Adapter)
- LEDs for Power, Ready, Status, WAN, PC, GSM
- Dial in mode or dial out mode only
- Call forward from GSM to VoIP and VoIP to GSM
- Dial Plan
- Password protection for both GSM dial in or dial out
- Retransmit GSM Caller ID to VoIP terminal
- Dynamic selection of codec
- Advanced jitter buffer
- Automatic traversal of NAT and firewall
- VLAN / Qos
- Echo cancellation for Speakerphone
- Comfort noise generation (CNG)
- Voice activity detection (VAD)
- Auto provisioning (requires auto provisioning server)
- Multi-language support: English and Chinese
- ITU: H.323 V4, H.225, H.235, H.245, H.450
- RFC 1889 - RTP/RTCP
- RFC 2327 - SDP
- RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
- RFC 2976 - SIP INFO Method
- RFC 3261 - SIP
- RFC 3264 - Offer/Answer model with SDP
- RFC 3515 - SIP REFER Method
- RFC 3842 - A Message Summary and Message Waiting Indicator
- RFC 3489 (STUN)- Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators
- RFC 3891 - SIP "Replaces" Header
- RFC 3892 - SIP Referred-By Mechanism
- Session Initiation Protocol Call Control Transfer
- Codec: G.711 (A/µ law), G.729A/B, G.723.1
- DTMF: RFC 2833, In-band DTMF, SIP INFO
- Web-base Management
- PPP over Ethernet (PPPoE)
- PPP Authentication Protocol (PAP)
- Internet Control Message Protocol (ICMP)
- TFTP Client
- Hyper Text Transfer Protocol (HTTP)
- Dynamic Host Configuration Protocol (DHCP)
- User account authentication using MD5