[asterisk-users] Connecting a GSM gateway to a FXO port

[Gordon Henderson]

On Fri, 17 Aug 2007, Hans Feringa wrote:

> I am trying to get a GSM gateway (Alpha Tech GSM Gateway Blue Gate Dual
> Band Analoog FXO) working with Asterisk.
> I had a working FXO configuration to a analog port of a small home 1/4
> ISDN pbx.
> I used this same configuration to connect a GSM Gateway that is supposed to
> be connected to the external(FXO) analog port of a pbx.
>
> I can get my configuration to dial the mobile number via the gateway, but
> Asterisk never detects that the remote side (the mobile phone) answered.

You're dialling the number of the SIM card in the gateway? Don't you want 
to dial another mobile number?

I have a GSM gateway here - (Telecom FM unit) I connected it into an FXO 
port, stuck a SIM card in it, and it "just worked". I can route outgoing 
calls through it and take incoming calls from it. As far as asterisk is 
concerned, it's just another FXO port and it treats it no different to the 
FXO port connected to my BT analog line...

Have you tried connecting a normal analogue phone to the unit and seeing 
what happens? Can you pick up the phone, get a dial-tone and dial a 
number? ... and does the phone ring when you call the SIM card number from 
another phone?

Maybe the SIM card you've put in it has a PIN that needs unlocking? (You 
did put a SIM card in, right?)

> When I use the r Dial option it will continue to signal a ringing tone to
> the local extension. When I leave out the r, I do get a connection but
> only for the duration of the configured number of seconds the Dial command
> is waiting for a connection. Because it did not detect the call being
> connected it will stop with a NOANSWER status effectively dropping the
> connection.
>
> I want to try two possible solutions:
> 1) I try to find out how to configure the GSM gateway behave in such a way
> that Asterisk correctly detects the call being connected.
> 2) I configure Asterisk to work with this device.
>
> The problem is that I looked thru the configuration options of the gateway
> and I could not find anything that made sense to me in relation to this
> prb.
>
> What (how) is the gateway supposed to signal back that the call is
> connected? If I know what is needed I can go back to my supplier with the
> right questions.

It's an analogue device - there is no way to (normally) send these signals 
back. Asterisk assumes the call is connected as soon as it's sent the last 
DTMF tone down the line... (AIUI)

> If the gateway can not be configured properly, I want to know how I could
> configure Asterisk to work around this problem and make it work anyway.

The gateway should act just like an ordinary analogue line from your 
telephone exchange, so if you can get asterisk to work with one of those, 
then you ought to be able to get it to work with the gateway.

At least thats how my unit works!

> This is the zapata.conf for the FXO port:
> [channels]
> ; hardware channels
> ; default
> usecallerid=yes
> hidecallerid=no
> callwaiting=no
> threewaycalling=yes
> transfer=yes
> echocancel=yes
> echotraining=yes
> immediate=no
> flash=100
> hanguponpolarityswitch=yes

Why? Does the device explicitly do this to signal a hangup?


> callprogress=yes
> progzone=nl
>
> busydetect=yes
> busycount=4
>
> language=nl
> ; define channels
> context=binnenkomend
> signalling=fxs_ks
> channel => 1 ; pstn attached to port 1
>
> This is the Dialplan fragment:
>
> exten => _87.,1,Wait(0.5)
> exten => _87.,n,Dial(Zap/1/${EXTEN:2},30,rtT)
> exten => _87.,n,Macro(fastbusy)
> exten => _87.,n,Hangup
> exten => _87.,102,Playback(tt-allbusy)
>
> [macro-fastbusy]
> exten => s,1,Answer
> exten => s,n,Wait,1
> exten => s,n,Playback(vm-isunavail)
> exten => s,n,Wait(3)
> exten => s,n,Hangup
>
>
>
>    -- Executing [870623027714 at home:1] Wait("SIP/kimura1-08236550", "0.5")
> in new stack
>    -- Executing [870623027714 at home:2] Dial("SIP/kimura1-08236550",
> "Zap/1/0623027714|30|rtT") in new stack
>    -- Called 1/0623027714
>    -- Nobody picked up in 30000 ms
>    -- Hungup 'Zap/1-1'
>    -- Executing [870623027714 at home:3] Macro("SIP/kimura1-08236550",
> "fastbusy") in new stack
>    -- Executing [s at macro-fastbusy:1] Answer("SIP/kimura1-08236550", "")
> in new stack
>    -- Executing [s at macro-fastbusy:2] Wait("SIP/kimura1-08236550", "1") in
> new stack
>    -- Executing [s at macro-fastbusy:3] Playback("SIP/kimura1-08236550",
> "vm-isunavail") in new stack
>    -- <SIP/kimura1-08236550> Playing 'vm-isunavail' (language 'nl')
>    -- Executing [s at macro-fastbusy:4] Wait("SIP/kimura1-08236550", "3") in
> new stack
>    -- Executing [s at macro-fastbusy:5] Hangup("SIP/kimura1-08236550", "")
> in new stack
>  == Spawn extension (macro-fastbusy, s, 5) exited non-zero on
> 'SIP/kimura1-08236550' in macro 'fastbusy'
>  == Spawn extension (macro-fastbusy, s, 5) exited non-zero on
> 'SIP/kimura1-08236550'
>
> Any help would be appreciated.
>
> Hans Feringa
>
>
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