Kanuri, Seshu (Company IT)
2005-May-12 08:27 UTC
[Asterisk-Users] GSM gateway for Asterisk
Folks! I am looking at a couple of models of Fixed GSM Gateways for the Purpose of VOIP connectivity and specifically to work with Asterisk. I found that these can be imported into USA for about $99.99 or about that. This is a one channel unit just like tellular, one of them has GPRS. FCT11M: 1)freq: GSM network,900/1800/1900Mhz, 2)provides reversal signal for payphone/billing 3)supports PBX and VOIP 4)for voice (no fax) 5)battery(optional) 6)can be used for the remote area where signal is weak. FCT11G: 1)freq: GSM network,900/1800/1900Mhz, 2)provides reversal signal for payphone/billing 3)supports PBX and VOIP 4)for voice and GPRS (no fax) 5)battery(optional) 6)can be used for the remote area where signal is weak. I am pasting an image of the network diagram here: Specifications in text are below. I would appreciate for any feedback of their usability. Seshu -------------------------------------------- Description: ----------- This unit can conveniently access to the available GSM system network. This system possesses such a high receiving sensitivity and a large transmitting power that it expandsthe effective coverage of the cellular network to a larger geographic area (upto 15 miles). The unit has been extensively used in the fixed access to the cellular network to solve the wired communications problems in the rural areas. It can also be used to develop fast radio public telephone services to satisfy the communications for the time being and work as the CO relay tosimplify the registration s and lower the cost. Furthermore it can meet the requirement of mobile communications onboard vehicles, ships, trains, etc. All these enlarge the number of the network subscribers considerably so that it can utilize the resources better. General Instructions How to link with a charger, Office PBX and VOIP Main functions -------------- Payphone Caller ID Pin number locked (Optional) Block prefix number (Optional) Support OfficePBX Support VOIP Office PBX VOIP Description/ Unit Specifications -------------------------------- UP MHz 890~915 1710~1755 1850~1910 WorkingFrequency DOWN MHz 935~960 1805~1850 1930~1990 Transmitting power dBm 33 Receiving sensitivity dBm -104 Atmosphere Kpa 86~106 Power Specifications Power mode: AC to DC a. Switch adaptor (without battery) 110-220V to 5V or 7-12V, 50/60Hz, 1.25A b. Switch adaptor (with Ni-MH battery) 110- 220V to 7.5V, 50/60Hz, 1.0A Backup battery: Standby: 20Hrs(Appr.) Continued Talking: 3Hrs(Appr.) Note: a. The battery will give the power when the normal power is off, and the battery power will be off when the normal power is On. B. The battery is for back up power only, It is not designed for normal power use. Quick Installation 1. Take off the cover of the SIM holder, then put in a SIM card into the holder. Receiving sensitivity dBm -104 2. Plug in a phone into the phone socket RJ-11 3. a. Install the antenna first, please screw the antenna tightly into the connector, and put the antenna in the purpose place. b. Connect the power, and put power switch ON. Power Specifications Power mode: AC to DC a. Switch adaptor (without battery) 110-220V to 5V or 7-12V, 50/60Hz, 1.25A b. Switch adaptor (with Ni-MH battery) 110- 220V to 7.5V, 50/60Hz, 1.0A Antenna information ------------------- Frequency range: A:890 960MHz B:1710 1880MHz Banwidth: A:70MHz B:170MHz Gain: 2.15dBi or 5.5dBi (optional) Impedance: 50? Max Power: 50W Connector Type: SMA Size: Longth:30cm , 60cm and 100cm (optional) Weight: 120g Other Specifications -------------------- Plastic cover: light blue or black Size 183mm 124mm 32mm(l\w h) G.Weight(complete set) 1.2Kg Circumstances: a temperature -20 ~50 b relative humidity 5%~95% Switches on the Box: ------------------- ANT ON OFF SET WORK LOAD RJ11 Antenna Power Switch for power Switch for set or work Set for factory only by now Please put the switch on work Load/USB for the factory only Rj11 for phone line Gain: 2.15dBi or 5.5dBi (optional) Impedance: 50? Max Power: 50W Connector Type: SMA Size: Longth:30cm , 60cm and 100cm (optional) Weight: 120g -------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited.
trixter http://www.0xdecafbad.com
2005-May-12 08:54 UTC
[Asterisk-Users] GSM gateway for Asterisk
On Thu, 2005-05-12 at 11:27 -0400, Kanuri, Seshu (Company IT) wrote:> Folks! > > I am looking at a couple of models of Fixed GSM Gateways for the Purpose of VOIP connectivity and specifically to work with Asterisk. I found that these can be imported into USA for about $99.99 or about that. This is a one channel unit just like tellular, one of them has GPRS.Something like this is similar to what I was asking about in a different thread, however a SIP/GSM protocol converter would be more ideal. Passively passing all data from the GSM network to the mobile and vice versa, thus removing any requirement for a SIM in the GSM device that gets installed. Basically the mobile would register through this becuase the signal strength is stronger, outbound calls would be routed to the PBX via SIP (or other, SIP would make more sense as its more universal), inbound GSM calls would be transparently bridged to the real mobile, all auth data would be passed so the mobile would have the SIM and perform as if it were directly connected to the GSM network. A SIP IM to GSM SMS bridge would also be really ideal. The ability for the SIP interface to cause a call to be initiated to the GSM network would also be ideal (granted this would require the phone to accept the auth data and reply accordingly, which could be a bit tricky, but if the GSM mobile user attempted to place a call it should work, although routing for that would have to exist on the GSM protocol converter itself rather than via the PBX. This would effectively turn any GSM phone into a pbx extension and/or SIP phone, with the ability for calls to come into that phone from the GSM network. I strongly feel that SIP would be better than trying to tie in an Abis interface into the PBX (those do exist commonly as a nanocell or picocell transceiver). Because the protocol converter does not need to decrypt via A5 the GSM calls, GSM MoU approval should not be required. In theory one could buy gsm transceiver boards and make their own device using an embedded (or just nanoitx and non embedded) solution, slap on some supported operating system and asterisk in the unit itself. Granted it would not always need to be a full on asterisk implementation since it does a very limited subset of features, but could be. That could incrase the SIP to all supported protocols. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050512/9fbc1370/attachment.pgp
I recently obtained a FCT-11M GSM-analog converter box. It arrived with no documentation. So I popped in a SIM chip, and connected the the RJ11 port to an FXO port on my Asterisk box. It worked smoothly right away for inbound and outbound calls in all respects. For about an hour. Then either spontaneously or due to some action I've been unable to identify, call supervision and other functions became flaky. First, I noticed inbound calls started malfunctioning. The Asterisk box answers, but no audio is heard on either end (the dialplan bridges to a SIP phone). Also, the call never ends. I can only knock it down by using CLI soft hangup or restarting Asterisk. Then outbound calls got wierd. It will dial out thru the GSM network to my cellphone, and audio is OK in both directions, but call termination fails if initiated from the remote GSM side. In that case, the box emits three short beeps, followed by a steady beep which is audible on the SIP phone to which the call is bridged. The channels don't hangup until the SIP phone causes it to. I was initially concerned that I had fried the FXO port by using an incompatible device, but I've ascertained that the port still works OK with a POTS line. I now suspect that the FCT-11M has been reconfigured somehow, since I obtained it and it was working. But I have no clue about how to examine it's configuration if possible at all. It has a USB (master) port but I don't know what it is for. Does anyone know if English documentation is available, or otherwise have any ideas on how to debug this? Much appreciate any insights.