Here are the steps how to connect GSM Gateway GoIP8 to Asterisk. Required features: the possibility to outgoing calls and receive incoming. For outgoing calls to have a choice of an arbitrary line.
1. GoiP 8 configuring
It is assumed that you have a new device out of the box, so the steps – as for factory settings.
Insert SIM card in slots, turn on the power. GoIP8 has 2 ports “LAN” and “PC”.
The «PC» port has a static address 192.168.8.1 by default and can be used to connect switches which can be connected to users’ PC (gateway supports DHCP server).
The «LAN» port can be used if your network is already running a DHCP server. In this case, the gateway will receive one of the addresses in your network. In order to know the IP-address your gateway has received, you can either:
- send to one of SIM-cards (with a positive balance) SMS or text info INFO. You’ll receive SMS with IP-address
- ask netwwork administrator
- enter the gateway IP address into the browser. The default username / password for authentication in the Gateway is admin / admin. You will be sent to “Status” page:
Next, go to:
Configurations -> Preferences and remove the “IVR checkmark (more about what each option does can be found in the instructions to the device, the link will be given at the end of the article).
On Configurations -> Call Settings page:
EndPoint type = SIP PhoneConfig Mode = Config By Line |
Next we should define configuration options for each line (X-is the line number):
Phone Number = 11XGateWay Prefix = *XSip Proxy = Asterisk_IPSip Registrar Server = Asterisk_IPAuthentication ID = 11XPassword = 11X |
GoIP8-call-settings
Next step is Configurations -> Call Divert page:
CID Forward Mode = Use CID as SIP Caller IDДалее для каждой линии прописываем настройки ( X-номер линии ):Forward to PSTN = EnabledDial Plan(VoIP to PSTN) = *X:-*XForward to VoIP = EnabledForward Number(PSTN To VoIP) =11X |
GoIP8 call divert page
Goip8 configuration to work with Asterisk can be considered complete :). Proceed to step
2. Configure Asterisk to work with GOIP
sip.conf:
Prescribes each channel individual user (X – channel number)
[11X]type=friendhost=dynamicusername=11Xsecret=11Xcontext=from-internaldisallow=allallow=alawallow=ulawqualify=yescanreinvite=nonat=no |
And Asterisk’s dialplan settings
extensions.conf
;исходящиеexten => _11X0XXXXXXXXX,1,Dial(SIP/${EXTEN:0:3}/*${EXTEN:2:1}${EXTEN:3})exten => _11X0XXXXXXXXX,n,Hangup();входящиеexten => _11Z,1,Answer()exten => _11Z,n,Playback(tt-weasels)exten => _11Z,n,Hangup() |
Dialing format:
11X0632223344 (where X is a channel number)
Done! Now you should check how calls go.
P.S. You can find GoIP8 manual here: http://dbltek.com/pdf/User_Manual_goip_series.pdf
