Without access to a cellular network, it is periodically impossible to organize telephony work, be it in a private home or a commercial enterprise. By connecting to the Internet, you can make the most profitable calls, which in some cases may not even be charged. Modern cellular companies take into account consumer demand, so they release starter packages and tariff plans with unlimited within the network. With their help, subscribers can enjoy communication via landline phones and IP options without spending a penny per call.
To use this option, you need to buy a VOIP-GSM gateway. Its main task is to connect the network and the cellular operator, which we will talk about in this article.
What it is?
While calls to landline phones from IP phones are very cheap, calls to mobile phones can cost the subscriber a tidy sum. This can be justified by the tariff plans of operators, which, based on the needs of users, try to provide favorable, unlimited tariffs for communication within the region, while other communication channels are charged quite high.
Since using cellular communications and IP options at the same time is costly and problematic, it is worth thinking about combining these two options. Now this has become possible thanks to the emergence of innovative devices - VOIP-GSM gateways. These are communication gadgets that can be connected to various mobile phones . With their help, the cost of calls within the network and cell phones within the city will be minimal, which is important both for private subscribers and for business owners.
The gateway simultaneously connects to IP/cellular operator networks thanks to several SIM card slots. Thanks to GSM channels, you can route calls in the following directions:
- From mobile devices to internal IP phones; IP network phones to cell phones;
- From IP phones to mobile operators;
- From phones of IP networks to other phones in the Russian Federation and abroad (using an Internet network);
Multi-settings make it possible to connect both dialing using GSM channels to numbers of cellular operators and calls via an IP network. Thus, a VOIP-GSM gateway is a great way to save money for calls between cities and countries.
Improved models collaborate with virtual PBXs, which guarantees expanded capabilities:
- support for SMS messages;
- call forwarding;
- installation and distribution of incoming/outgoing calls;
- working with SIP accounts and SIP lines;
- sending USSD combinations;
- telephone emulation;
- setting up black lists of unwanted subscribers;
- placing calls on hold/hold.
The gateways can contain up to a couple of dozen SIM cards at the same time, which is very convenient for commercial use, when collaborating with clients of different cellular operators. Of course, such models are an order of magnitude more expensive than “home” options with one or two SIM card slots.
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Which GSM gateway to choose
It all depends on for what purpose and for what load the device is purchased. So, if you are planning to install it in a small office or country cottage without a mini-PBX, then it would be reasonable to take a bridge with one SIM card and several FXO ports for connecting stationary devices directly.
For mini-PBX and IP telephony with the ability to organize video conferencing, install faxes and other useful options, you need a multi-channel gateway with VOIP and GSM, where each channel has its own SIM card and the more connection points, the more powerful the device should be.
Advantage of remote access to SIM cards:
- centralized management of SIM cards,
- their dynamic distribution,
- no need to replace or reinstall SIM cards in the device,
- hot swapping of SIM cards without any interruptions in service.
For example, a GoIP gateway at a remote site will no longer require replacing SIM cards in the device or reinstalling them. A group of GoIP gateways can be managed remotely efficiently and cost-effectively using a SIM bank. The main advantages of using a SIM bank are hot-swappable SIM cards in gateway channels (gateways) geographically located from the SIM bank itself and other gateways to implement the function of moving SIM cards between GSM operator base stations, autonomous operation with support from 32 to 128 independent SIM card connections.
Terminology
First, let's take a short excursion into terminology:
IP (Internet Protocol) refers to all data transmission over the Internet.
- IP telephony - all calls and video communications over the Internet.
- VoIP (Voice over IP) - transmission of any voice data over the Internet: regular audio call, video call, voice message, webinars, films, other voice media. That is, it is IP telephony + media.
- SIP (Session Initiation Protocol) is one specific protocol that enables IP telephony. In addition to SIP (which is the most popular), there are other protocols: H.323, H.248, MGCP, SIGTRAN, Jingle, SCTP... That is, a protocol is a specific technology used for communication, and IP telephony is, in general, the ability to transmit voice over the Internet. Here's an example: if IP telephony is a smartphone (generalized name), then SIP is Samsung (a specific type).
- GSM gateway is a general name for devices that need to redirect media data from IP (internet) to GSM (mobile communications).
When we call from a mobile phone, we use the GSM - mobile network. When we call via Skype, we use one of the IP protocols - the Internet. A gateway helps to forward data from one network to another.
Popular gateways
As examples, we will look at the three most popular models of VOIP-GSM gateways.
Yeastar NeoGate TG400
The list of popular VOIP-GSM gateways includes the Yeastar NeoGate TG400 model. The gateway is aimed at office applications where you need to make frequent calls to mobile phones. Especially for this, the device was equipped with four slots for SIM cards. The gateway can work with SIP providers and can connect to office PBXs .
To improve the quality of communication with GSM networks, it is possible to connect external antennas. Work with GSM networks in the Yeastar NeoGate TG400 gateway is presented in full - the device can work with SMS mailings, receive text messages, send USSD commands and inform about the balance on the numbers used. If necessary, installed SIM cards can be replaced without rebooting the device.
Yeastar TG100
This miniature VOIP-GSM gateway is equipped with one GSM module and a built-in SIP proxy for routing calls from IP phones. A LAN port is provided for connecting to external equipment. The device can route both incoming and outgoing calls, allows you to check your balance and warn about its decline, can suppress echoes and work with SMS messages via a web interface. SMS messaging is also supported.
GoIP 1
The miniature GSM-SIP gateway GoIP 1 can be used both at home and in a small office . It is equipped with one SIM card slot, and IP phones are connected via IP PBX or switches. Also, the use of soft phones is allowed. Thanks to its small size, affordable price and good performance, the device is an ideal solution for routing calls between IP phones and GSM networks.
This VOIP-GSM gateway is available in several modifications, differing in the number of installed SIM cards.
How to save money using a GSM gateway
As already mentioned, there are many options for operating the bridge, but the main savings are observed when using the simplest scheme. Suppose an office employee with a mini-PBX connected to a city line needs to call a companion on his mobile phone.
A call from an office PBX first arrives at a city exchange, is converted there and sent to the cellular operator, and from it to the subscriber; in this chain, the most expensive is the transition from landline to mobile. If a gateway is installed, then the “freeloader” in the form of the city station disappears. An office employee directly calls the recipient using the internal tariffs of the cellular operator.
In conclusion, I would like to note that disputes surrounding the legality of installing such equipment periodically arise; some operators even try to block the owners of gateways, but at the moment there are no laws prohibiting the use of bridges, which means it makes sense to install this convenient, and most importantly, useful device.
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OpenVox GSM gateway as a complete replacement for PBX
As you know, SIP is gaining more and more popularity and it’s no secret that you can optimize telephony costs by choosing the right SIP provider.
But, unfortunately, the above statement is true only for large cities in Russia and the CIS. But what to do in cases where SIP is not available? Use PRI? Quite expensive, especially for a small company with a small number of calls, and again, not available everywhere. Use analog telephony the old fashioned way? Certainly not. The answer seems quite obvious: a GSM gateway will solve all your problems.
At first glance, everything is quite simple to implement, but upon more detailed consideration, it will become clear that the gateway itself has little functionality and is only a kind of GSM to SIP converter, and to implement even the most unreasonable, but still “wants” you need PBX (for example IVR, routing, queues). To minimize the cost process, let's say that this is our favorite open-source Asterisk. If we are talking about completely global savings, then we can immediately say: “Yeah, what about the server for Asterisk? It’s not free either!” (Although, in my opinion, a thrifty admin will always have hardware for Aster :))
Now imagine that everything written above can be implemented using just one GSM gateway? Incredible? Well, please, under the cat.
A couple of months ago, grigly wrote an article about the new GSM gateways of the Openvox company, which ended with a quite optimistic and action-motivating phrase: “Colleagues, it has full-fledged Asterisk, I wouldn’t even be surprised if it’s possible to set up IVR and queues on it, and use it as full-fledged automatic telephone exchange.”
But really? Is it possible to implement PBX functions using a gateway?
Colleagues, I’ll immediately say that this solution is suitable for a small company, say with a staff of 10 employees.
So, first, let’s evaluate what PBX functions a small company needs? In my opinion, these are queues, IVR, call transfer.
Let's create sip peers in the gateway web interface:
Go to the SIP → SIP Endpoints tab, click “Add New SIP Endpoint”
Name = 777 User Name = 777 Password = YourStrongPasssword123 Registration = Endpoint registers with this gateway Hostname or IP Address = dynamic Transport = UDP NAT Traversal = Yes
Click “Save”
Next, by analogy, we create the required number of SIP peers.
Now let's enable SSH access in the web interface. To do this, go to the System → Login Settings tab and set SSH Login Settings to “Enable ON”
It is worth noting that the port for SSH in gateway 12345 cannot be changed, which was apparently done for security purposes.
By default, the admin user does not have root rights; in order to become root, you need to change the user to super, the password is super.
So, we go to the gateway via SSH:
ssh -p 12345 [email protected] Password: admin
After you get to the gateway, edit the context of the extra-channels.conf configuration file:
vim /etc/asterisk/extra-channels.conf:
Edit the context for an incoming call from SIM1:
;
Span 1: opvxg4xx/0/1 “OpenVox G400P GSM/CDMA PCI Card 0” AMI/CCS group=1 context= ivr
signaling = gsm vol=70 mic=1 dacgain=-15 adcgain=-3 debugat=off smscodec=utf -8 ;hwdtmfdet=1 anonymouscall=off call_waiting=off band= dialprefix= switchtype=SIMCOM_SIM840W channel => 1
Now let's create a new ivr context.
vim /etc/asterisk/extensions_custom.conf
Create a new ivr context:
[ivr] exten => s,1,Answer() same => n,Background(press-1&or&press-2) same => n,WaitExten() exten => 1,1,Playback(wait_connection) same => n, Dial(SIP/777) same => n,Hangup()
exten => 2,1,Playback(wait_connection) same => n,Dial(SIP/888) same => n,Hangup()
The dialplan in the example is the most primitive, intended for demonstration; in production, of course, protection against incorrectly entered numbers must be implemented.
It's worth noting that Openvox only supports sounds for IVR in GSM format. I had sounds in .wav format, which I transcoded using the system:
sox input.wav -r 8000 -c1 output.gsm lowpass 4000 compand 0.02,0.05 -60,-60,-30,-10,-20,-8,-5,-8,-2,-8 -8 - 7 0.05
By default, gsm modules are not loaded in Openvox; you need to force them into the CLI:
Openvox-Wireless-Gateway*CLI> module load codec_gsm.so Openvox-Wireless-Gateway*CLI> module load format_gsm.so
In order not to load gsm every time after rebooting the gateway, I recommend adding these two lines to the /etc/asterisk/modules.conf file
Now let's move on to configuring queues:
Go to the gateway web interface, Routing → Groups tab. Create a new group by clicking the New Group button:
Group Name = YourGroupName Type = SIP Policy = select the calling strategy that suits you Members = check the checkboxes opposite the numbers that we want to include in the group:
Now go to the Routing → Call Routing Rules tab, click New Call Routing Rule:
Routing name = YourRoutingName Call Comes in From = for example, gsm-2 (that is, all calls from sim 2 will be redirected to our queue) Send Call Through = select the newly created group from the drop-down list.
Another important feature is the ability to transfer calls. To our great joy, this is already provided by the Openvox developers. Let's open feauteres.conf:
vim /etc/asterisk/features.conf
And we see the following line there:
blindxfer => #1 ; Blind transfer
That is, pressing #1 will transfer the call; in principle, you can select any prefix for transfer; for convenience, I changed it to *1.
Conclusion
Using the undocumented capabilities of this device, the result is the following cost estimate for a typical small business:
1) 10 IP phones ~ 25 thousand rubles. (not the cheapest devices, but with support for HD codecs, headsets, and probably even PoE)
2) OpenVox VS-GW1200-4G ~ 15 thousand rubles
3) If you need to connect an external analog line and, say, a fax, then you can purchase an inexpensive FXO / FXS gateway, for example Granstream HT-503 ~ 2.5 thousand rubles.
4) As mentioned above, the gateway can both register end SIP devices on itself and register itself with the PBX, so there will be no problems connecting to a SIP provider
Total, full-fledged VOIP in the office, with unlimited trunks, queues and blackjack for less than 45,000 rubles!
The gateway does not have hard drives and uses only high-quality components (we could see this in the previous article about OpenVox), so it is an extremely reliable solution.
Moreover, this solution does not require any additional ones. licenses, subscriptions to technical. support and so on, so the total cost of ownership is calculated quite quickly and you can forget about all sorts of pitfalls with licensing.
