MOBILE DID TRUNK & EXTENSION SETTINGS FOR ASTERISK/FREEPBX/FREESWITCH/3CX

After you have received your welcome notice from GTI, follow these steps. These are BASIC settings to get your PBX system up and running!

STEP #1  ADD (Whitelist) THESE IP IN YOUR SERVER FIREWALL

99.198.122.166
198.143.185.83
198.143.185.82

96.127.174.38
37.18.129.170
37.18.129.171
37.18.129.172
37.18.129.173

STEP #2  This Trunk allow GTIGLOBAL to send traffic to your server and is only use for those reasons...  add the following information below:  (This example uses IP Authentication for your Trunk) What is SIP Trunking? ⋆ Sangoma

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SIP Trunk:

host=mobiledid.mvnoserver.com
context=from-trunk    or   context=from-pstn-e164-us
type=friend
port=50906

For FreePBX Trunk you may use the following:

type=friend
host=mobiledid.mvnoserver.com
qualify=200
disallow=all
allow=ulaw
allow=g729
allow=alaw
dtmfmode=rfc2833
nat=auto_force_rport
directmedia=no
;insecure=port,invite
send_pai=yes

port=50906

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STEP #3  Add the DID Number for SIM in your FreePBX Inbound make sure to use the 11 digits.Cloud, computing, data, platform, process, server, virtual icon

STEP #4  Create PJSIP/SIP Extension  In your PBX (use same extension number and password as in the GTI Portal). Assign/route the DID to the PJSIP extension in your PBX.

STEP #5  Test the DID to see if it is reaching your PBX, call the DID Number Associated with the SIM (make sure you call from a line that is not connected to your network for a real live test).

STEP #6  Install the SIM in your phone https://support.gtivoice.com/knowledgebase.php?article=58

STEP #7   Using the Cellphone, make a call to see if you can make call out from your PBX. You can dial *97 or your voicemail feature code if it available assuming you setup voicemail for the extension. Check your CDR for the call results.

If you need further help, please open a support ticket.

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