Configuracion de Gateway GSM GOIP con Asterisk

Este aparito llamado GOIP es un gateway GSM su función no es mas que comunicar la red de teléfono movil con la red de telefonía VOIP.  Es decir que tomamos un GOIP le insertamos la tarjeta sim de nuestro celular lo configuramos con nuestra PBX (Asterisk,Elastix,Trixbox)  y ya podríamos usarlo como troncal  para nuestro central para hacer y recibir llamadas.

ASTERISK SETUP

Create Asterisk extension to be used by goip

Set strong password for extension

Create Trunk for Goip with following settings:-

Dial Plan:

X. (accept anything from outbound rules)

Trunk Name:  goip

PEER Details:

username=goip (can be anything, doesn’t really matter)

type=peer

qualify=yes

port=5060 (This is optional unless changed from 5060 but must be same as set in “Advance settings” in GOIP. If multiple goip used, each should have separate port)

nat=no

insecure=very

host=192.168.1.202 (Static IP address assigned in GOIP)

context=from-pstn

canreinvite=no

User Context 0412XXXXXX

User Details:

username=goip (same as above)

type=user

port=5060 (same as above)

nat=no

insecure=very

host=192.168.1.202

context=from-pstn

canreinvite=no

I have found that this section (User Details) is not really necessary, all can be omitted as long as “User Context” is set and it will still work.

Register String : (Not Necessary)

Much valuable information about setting up and securing Elastix can be found here : http://www.sunshinenetworks.com.au/how-to.html

GOIP SETUP

Lan Port : set to Static IP

IP Address : same IP address used in Asterisk Trunk setup

Default Route : Modem IP address

Primary DNS : 208.67.222.222 is opendns. You can use this or the modem ip or your ISP DNS

Endpoint Type : SIP Phone

Config Mode : Single Server Mode

Phone Number : This MUST be Asterisk extension number

Display Name : can be anything

SIP Proxy : MUST be Asterisk server IP address

SIP Register Server : MUST be Asterisk server IP address

Authentication ID : MUST be asterisk extension number

Password : MUST be asterisk extension password

Call forward type : should be “Not Forward” otherwise incoming calls will go to “Call Forward Number”

Ensure “Local Signaling Port” under  “Advanced Settings” is set to same port as asterisk trunk if this was changed in Asterisk

Ensure “RTP Port range is consistent with your Asterisk setup (default asterisk is 10000 to 20000)

Set Audio Codec Preference and order to suit your setup

Forward to PSTN : enabled

Forward Number : This is where the call will go if an internal asterisk caller rings the extension number. I have it going to my mobile number.

Forward to Voip : enabled

Forward Number : this is where incoming calls to the goip mobile number will go. This can be an asterisk extension or a call group. I have it going to call group 6XX

Set CID Forward Mode to “Use CID as SIP Caller ID” if you want caller ID passed to Asterisk. If you enter something in “CID Prefix”, this will appear with caller ID in asterisk log.

Save changes

If all is operating correctly, Line Status should show “Login” and GSM Status should show “Login”. If not, try reboot goip. I have also found the settings don’t always take in Firefox. It is safer to use IE to do setup.

Only the items referred to above are necessary to get goip working. If I have not mentioned a setup option, the default settings should be adequate.

With this setup Goip will do three things:-

-                      External call to Goip mobile number will ring Asterisk extension or ring group set at “Forward Number (PSTN to Voip)”

-                      Internal call to Goip extension number will forward call to number set at “Forward Number (Voip to PSTN)” using Goip sim.

-                      Asterisk call using Goip Trunk will dial number using GOIP sim.

[from-internal-custom]

exten => _905.,1,Dial(SIP/700/${EXTEN:3})
exten => _905.,3,Hangup()