Asterisk, GSM-FXO gateway example of usage

Question

Can anyone explain where to use GSM-FXO gateway. I understand where to use GSM-FXS - for example to connect analogue telephone to gsm gateway. But do we use GSM-FXO to connect to two telephone stations?

Answer 1

I have to admit that I've never encountered such equipment before, so my answer will be based only on wild speculation :)

As you surely know, FXO defines the interface facing the carrier's network. FXO, unlike FXS doesn't provide a dialtone, line voltage etc., but instead can dial numbers into the network.

Now, depending on the featureset of the GW, the most plausable scenario would be adding mobile users into hunt groups or follow-me scripts for PBXs equipped only with analog extension ports

Other, more complicated ideas come to mind (i.e. termination of mobile traffic), but those make much less sense compared to the previous one.

Answer 2

A GSM FXO is no different to an analogue FXO in terms of operation, the difference is only in the medium for which it connects.

I have one on my network which I use to connect my Analouge BT phone line into my Asterisk PBX service, so that when inbound calls come into my fixed UK land line, they get picked up by asterisk and handled appropriately.

In my case, I use the FXO (More commonly known as a PABT unit) to watch for a ring signal, then it passes that ring signal to Asterisk in the form of a standard SIP VOIP communication.

As far as Asterisk is concerned, it just looks like another extension on the network, but I have this one tagged to go into a different dial plan.

My Dial plan, then looks at the incoming number, and compares it to certain things, such as my spam black list and caller ID lookup database.

Depending on the inbound number, the call is dropped, or I play monkies, re-route to a VPS etc.

If I decide to allow the call through, I use asterisk AGI, to change the inbound number into a name, which then gets displayed on my VOIP phones.

I also have it working the other way.

If I dial a given prefix on my asterisk exchange, I can send a number out on my BT Land line. Other prefixes allow me to select from 2 or 3 different voip providers.

The unit I Have is a Linksys SPA3000. I'll not describe the actual setup on the device, as that's specific to that particular brand, but as far as asterisk is concerned:

In extensions.conf i have

;--------------- External calls coming in from BT line enter here -----------------------------------------
exten => BTLINENUMBER,1,Verbose(2,Starting default dialplan for BTLINENUMBER extension)
  same => n,GotoIf($["${CALLERID(num)}" = ""]?nocid)

  ; We end up here if caller ID was presented
  same => n,AGI(${inboundLog})
  same => n,Dial(${Office1}&${Attic}&${PetersPC}, 15, x)
  same => n,VoiceMail(${GlobalVM},u)
  same => n,Hangup()

  ; We end up here if caller ID was not presented
  same => n(nocid),Playback(silence/4)
  same => n,AGI(${withheldlog})
  same => n,Read(menukey,"custom/pbx-witheld",1)
  same => n,GotoIf($["${menukey}" = "1"]?leavemessage)
  same => n,GotoIf($["${menukey}" = "2"]?bookcall)
  same => n,Playback(vm-goodbye);
  same => n,Hangup()

  ; Witheld caller can leave a voice message
  same => n(leavemessage),Verbose(2,Leave a message here)
  same => n,Goto(voicemailwitheld,1,1)
  same => n,Hangup();

  ; Witheld caller can use call booking service
  same => n(bookcall),Verbose(2,book a call here)
  same => n,Playback(custom/no-booking-service)
  same => n,Hangup();

[voicemailglobal]
exten => 1,1,Verbose(2,Starting voicemail global)
  same => n,VoiceMail(${GlobalVM},u)
  same => n,Hangup()

[voicemailpeter]
exten => 1,1,Verbose(2,Starting voicemail peter)
  same => n,VoiceMail(${PetersVM},u)
  same => n,Hangup()

[voicemailsam]
exten => 1,1,Verbose(2,Starting voicemail sam)
  same => n,VoiceMail(${SamsVM},u)
  same => n,Hangup()

[voicemailwitheld]
exten => 1,1,Verbose(2,Starting voicemail global)
  same => n,VoiceMail(${WitheldVM},b)
  same => n,Hangup()

In sip.conf I have the following entry

[PSTN]
type=peer
context=default
canreinvite=no
nat=no
host=192.168.17.99
port=5061
callerid="BTPHONENUMBER"
dtmfmode=rfc2833

As you can see it's set as a non authenticating peer connection. Unlike most normal connections for handset's which you might typically authenticate then re-register periodically.

You can see that the context is set to the context shown above from the extensions file, and all that's then left to do is to configure your FXO device to forward all calls without answering them to your asterisk box, using the PSTN information.

In my case this involves sending the caller ID (Which I've changed to BTPHONENUMBER to protect it from spam calls) and making sure that the FXO unit passes the IP and port across.

How you connect your FXO to asterisk is entirely dependent on what it's model is, but connected up in the same manner as mine should allow you to make and receive calls on a GSM mobile network.