Hi, I’m really new using grandstream gateways like GXW 40XX (FXO) and GXW 42XX(FXS). At work, I have a task using these equipments and that is to interconnect analog channels, 4 wires(2-TX and 2-RX) with those gateways. I have four Gateways:2xGXW4008 and 2xGXW4224. The thing is the analog channels are audio channels: they don’t use signaling like E&M and they behave like they are passing audio “all the time”. I had looked in the network and I found something called “hotline”, but I don’t know if that works here.We also have some cards that use that type of signaling, but they use 6-wires(a pair more for signaling data E&M).The goal is to achieve, using the 4-wires card, connecting to a gateway, each pair of wires for TX and RX audio. The other thing is that those cards(4-wire) are connected using the wires with other cards that processes the audio. We want, using 2 gateways, to pass the channels , travel using VoIP and then down the audio to the other card. Is it possible that the gateways can pickup automatically and pass the audio to another channel?
The interface connection on the GXW series of product is an RJ11 and only uses Tip & Ring, the inner two contacts in the RJ11 connector. The device is used to provide a POTS line to an analog phone from a Ethernet input using the SIP protocol. A common use of the device is to provide telephone service to hotel rooms equipped with analog phones from a IP-PBX system.
It does not accommodate E&M signaling nor will it pass “audio channels” without some basic PSTN telephone signalling - ring, off-hook, current loop, etc.
If the device to which you want to connect the GXW is a telephony device, perhaps there is more to what can be done, but as it stands with the current description, I do not think the GXW will do what you need.
Thanks for help.But still I´m asking something related with my chances(less than 1% ): If I " pick up" the phone, so to speak, is possible than it automatically calls another phone? Reading the docs, I found unconditional call forwarding(I’ve read that some people called it hotline) but never tested. Adding to that, how long the call can last ( I mean, eternally!) in order to establish a forever channel audio? We maybe became “creative” with the PSTN signalling.
Unconditional Call Forwarding (UCF) and hotline (auto-dial) are not the same.
UCF is when a call comes into the device, the device will forward to some other number. It will never pass the call to the endpoint connected to the device.
Hotline is when the phone connected to the ATA goes off-hook, the ATA will dial a pre-programmed number automatically with no further user input.
The line will stay alive as long as neither end hangs up or the circuit remains uninterrupted,
Still, those products (GXW4224 or GXW4008) support hotline? The question was never answered
Yes, they support off-hook auto-dial.
Alright then. Thanks! I’let you know if I progress at work.
I have a question, also referred with this topic. We use a modem card, so the question is: The Gateways support modem connections? The card will connect to the Gateway through lines and in other Gateway we have also a card connected the same. Both cards are needed to establish a serial communication, like many years ago . Also, is it possible than modem(serial communication) is supported over IP? Thanks in advance, I know my questions may sound odd or rare, but please help.
Let’s just say that while it may be possible, consider that it will depend on the modem type and the dependability of the internet connection,. The ATA/gateways are taking an analog modem feed and then converting it using a voice codec. The other end must then take the IP packets and convert these back into an analog signal for use by the opposing modem.
However, as you are asking about serial, then perhaps you are really in need of a Ethernet (TCP/UDP) to serial adapter. If you are trying to control something that requires a serial input, then there are a number of devices that can do and avoid the double conversion. Take a look at Comtrol.com.
I got a mistake. Serial has nothing to do here. Everything is just about modems. Modem over IP is the deal. Still, I’m wondering how to do it using the gateways. Using Asterisk I can bridge two VoIP Channels (SIP Channels), not to mention I’m still planning to use the hotline ability for the “channel audio”. Let me add this: those are analog channels. See below…
I will explain the problem:
Here at work we use private trunking system called TaitNet, based in the standard MPT-1327. So you may say it’s an old stuff, but we offer services to our clients and also we are migrating to a DMR platform. The MPT1327 system uses cards four-wires, but the description mentions analogue I/O card. Let me add, our basestations also have a " special card" for the signaling part of the basestation channels and to pass the call to a specific analog channel. That card is esentially a modem card .The idea: digitalize the channels, including the modem. Thats why I asked the channels audio at the very beginning. Additionally, every card has a counterpart card, meaning, that every channel has a corresponding part or card at the core of the network.
They are four-wires, meaning two wires for TX and two for RX. The RX in one card gets connected to the TX in the other card and the procedure is repeated, now with TX -card-one with RX-card-two. As you see, no signaling E&M is applied here. We also have other cards, telephone cards that uses E&M, but 6-wire: 2 TX,2RX and 2E&M, meaning the signaling is out of the band.
We want to use two gateways here. One for the conversion to IP and the other for reversing that process
Why would you not talk to TAIT?
The issue is the same - the transitions from your RF trunk network to a telephony interface and then back at the other end. You have to provide ring and then other side has to answer. These are not typical modems, but rather modems that use a voice codec with telephony type signalling.and interfacing at each end. The number of wires is inconsequential unless you can get into a tip and ring (2) to feed the gateway at each end.
Please refer to the following -