Elastix – PBX:

1.Select – Trunks – Add Sip Trunk

General Settings

Trunk Description: PSTN

Outbound Caller ID: YOUR PHONE NUMBER

CID Options: Allow any CID

Maximum Channels: 1

Outgoing Settings

Trunk Name: PSTN

PEER Details

context=from-trunk
username=300 (USE THE SAME ID ON YOUR GW)
secret= YOUR PASSWORD (USE THE SAME PASSWORD ON YOUR GW)
host=dynamic
port=5061
srvlookup=yes
insecure=port,invite
canreinvite=no
dtmfmode=rfc2833
t38pt_udptl=yes
nat=no
qualify=yes
type=peer
disallow=all
allow=alaw&ulaw
sendrpid=yes

Submit and apply!

2.Select Outbound Routes – Add Route

Route Name: 9_outside

Music On Hold: Default

Dial Patterns: 9|.

Trunk Sequence: Select the SIP TRUNK that you have created above.

Submit and apply!

3.Select Inbound Routes – Add Incoming Route

Description: ADD A NAME

Music On Hold: default

Privacy Manager: No

Language: en

Source: none

Set Destination

Extensions: Select on which extension do you want the call to be routed.

Submit and apply!

4.SPA 3000/3102:

Proxy and Registration:

Proxy: ASTERISK IP

Use Outbound Proxy: no

Outbound Proxy:

Use OB Proxy In Dialog: no

Register: yes

Make Call Without Reg: yes

Register Expires: 3600 (On LINE 1 change it to 300)

Ans Call Without Reg: yes

Use DNS SRV: no

DNS SRV Auto Prefix: no

Proxy Fallback Intvl: 3600

Proxy Redundancy Method: Normal

Subscriber Information

Display Name: NAME OF YOUR SIP TRUNK THAT YOU CREATED PREVIOUSLY. In our e.g. PSTN.

User ID: ID OF THE TRUNK THAT YOU PREVIOUSLY CREATED. In our e.g. 300.

Password: THE PASSWORD THAT YOU PREVIOUSLY ADDED.

Use Auth ID:

Auth ID:

Mini Certificate:

SRTP Private Key:

If something is not mentioned, leave it blank. You should be able to make calls (9number#) and receive calls to the extension that you have created and declared in both asterisk and SPA/LINE 1.