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

[Hans Feringa]

Thanks for your response. My answers below.

> 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?

No I am dialing out to another mobile number (bad english on my part).
>
> 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?
Works with an analogue phone. ..
Dialing out from Asterisk to a mobile number works, however picking up the
mobile phone and establishing the connection is not detected by Asterisk.
(see below)
>
> Maybe the SIM card you've put in it has a PIN that needs unlocking? (You
> did put a SIM card in, right?)
>
No pin required ..

>> 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)
>
That is what I understood at first. I could not understand it's different
behaviour from the previous connection to the pbx. Maybe it has something
to do with the current that flows when it is off hook?

>> 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.
>
My thoughts exactly.
> 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?
>
Hmm, a leftover of my testing the hangup situation on the pbx. I guess I
should remove that, because it started working by configuring busydetect.
>
>> 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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>
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