1. GOIP Configuration.
It is assumed that the gateway has the factory configuration.
Insert the SIM cards into the gateway. Power it on.
Now, access the gateway's web interface. To do this, connect the GOIP to the local network using an Ethernet cable. There are two ways to do this:
1) Via the “PC” port. By default, it has the IP address 192.168.8.1
2) Via the “LAN” port, if a DHCP server is running on the network. By default, DHCP (dynamic IP address) is configured on the “LAN” port. To find out which IP address was assigned, you can either:
a) Send an SMS to the gateway (to the SIM card with a positive balance) with the text info or INFO. You will receive an SMS response containing the IP address.
b) If you know Chinese – call the gateway, and after the connection is established, dial *00.
c) Ask the network administrator.
Next, enter the gateway's IP address into your browser. By default, the login/password for gateway authorization is admin/admin. You will land on the initial “Status” page.
Next, go to:
Configurations -> Preferences and uncheck IVR. This will disable the voice menu on the device.
You can leave Auto Reboot enabled for automatic gateway rebooting; apparently, this is how the manufacturer combats bugs in the device.
Configurations -> Call Settings
EndPoint type = SIP Phone
Config Mode = Single Server Mode
Next, specify the internal extension on Asterisk to which the device will register.
Phone Number = 11X
Sip Proxy = Asterisk_IP
Sip Registrar Server = Asterisk_IP
Authentication ID = 11X
Password = 11X
Configurations -> Call Divert
Enable caller ID forwarding:
CID Forward Mode = Use CID as SIP Caller ID
Forward to PSTN = Enabled
Forward to VoIP = Enabled
Forward Number(PSTN To VoIP) =11X
Configuring Elastix (Asterisk) to work with GOIP.
1. Create an internal extension (Extension) under which the device will register on our server.
2. Create a trunk (Trunk) and specify the Outgoing Settings.
Trunk Name: GOIP_out
PEER Details:
host=192.168.8.1 – GOIP address
context=from-internal
type=peer
port=5060
nat=no
qualify=yes
canreinvite=no
insecure=very
dtmfmode=auto
disallow=all
allow=alaw&ulaw
3. Specify the Incoming Settings.
USER Context: GOIP_in
USER Details:
host=192.168.8.1 – GOIP address
type=friend
context=from-trunk
################# Second Article
Configuring GOIP.
It is assumed that the gateway has the factory configuration.
Insert the SIM cards into the gateway. Turn on the power.
Now, access the gateway's web interface. To do this, connect the GOIP to the local network using an Ethernet cable. There are two ways:
1) Via the “PC” port. By default, it has the IP address 192.168.8.1
2) Via the “LAN” port, if a DHCP server is running in the network. By default, DHCP (dynamic IP address) is configured on the “LAN” port. To find out which IP address was assigned, you need to either:
a) send an SMS to the gateway (to the SIM card with a positive balance) with the text info or INFO. You will receive an SMS with the IP address in response.
b) for those who know Chinese – call the gateway, and after connecting, dial *00.
c) ask the network administrator.
Next, enter the gateway's IP address in the browser. By default, the login/password for authorization on the gateway is admin/admin. You will land on the initial “Status” page.
Далее заходим:
Configurations -> Preferences и убираем галочку IVR.
Configurations -> Call Settings EndPoint type = SIP Phone Config Mode = Config By Line Далее для каждой линии прописываем настройки ( X-номер линии ) Phone Number = 11X GateWay Prefix = *X Sip Proxy = Asterisk_IP Sip Registrar Server = Asterisk_IP Authentication ID = 11X Password = 11X
Configurations -> Call Divert CID Forward Mode = Use CID as SIP Caller ID Далее для каждой линии прописываем настройки ( X-номер линии ): Forward to PSTN = Enabled Dial Plan(VoIP to PSTN) = *X:-*X Forward to VoIP = Enabled Forward Number(PSTN To VoIP) =11X
Configuring Asterisk to work with GOIP.
sip.conf:
Specify each line for registration on Asterisk (X is the channel number).
[11X] type=friend host=dynamic username=11X secret=11X context=from-internal disallow=all allow=ulaw allow=alaw qualify=yes canreinvite=no insecure=very nat=no
Configure the Asterisk dialplan.
extensions.conf
;outgoing
exten => _11X0XXXXXXXXX,1,Dial(SIP/${EXTEN:0:3}/*${EXTEN:2:1}${EXTEN:3})
exten => _11X0XXXXXXXXX,n,Hangup()
;incoming
exten => _11Z,1,Answer()
exten => _11Z,n,Playback(demo-congrats)
exten => _11Z,n,Hangup()