Elastix is an Open Source Unified Communications Software. SIP Trunk configuration instructions below apply to the following Elastix versions:
Documentation is provided for scenario where Elastix server uses Static IP address on the public Internet and when Dynamic IP address is used.
To begin SIP Trunk configuration open PBX Configuration:
PBX
in top toolbar.Next follow "Static IP addres" or "Dynamic IP address" section below.
PBX Configuration
in left toolbar.Next follow "Static IP addres" or "Dynamic IP address" section below.
Static IP address (a.b.c.d
in our example above) of your Elastix server will be added to GoTrunk service IP ACL (Access Control List) and outbound calls coming from that IP address will be accepted without requiring any further authentication (SIP username and password). This is the most efficient way of authenticating SIP calls.
Inbound calls to one of Telephone Numbers on your GoTrunk account will be sent directly to Elastix public IP address. Since the calls will be coming from known peer
(IP address of SIP Trunking service q.x.y.z
in our example above) Elastix will accept them without requiring any further authentication.
To configure Elastix server to work with GoTrunk SIP trunk using IP authentication the following changes are required:
Follow steps below to add SIP Trunk:
1. Select Trunks
.
2. Click Add SIP Trunk
button.
3. Enter name of the trunk as gotrunk
4. Enter the following into PEER Details
field (replace eu.st.ssl7.net
with amn.st.ssl7.net
if you want to use North America POP):
type=peer host=eu.st.ssl7.net context=from-trunk
5. Click Submit Changes
button.
Next follow "Routing configuration" instructions below.
For outbound calls from Elastix to GoTrunk SIP Credentials (SIP username and password) authentication is used.
For inbound calls to one of Telephone Numbers on your GoTrunk account to work Elastix needs to Register with GoTrunk service (and periodically refresh registration in case IP address changes). Calls will be sent to IP address which was sent in the most recent Elastix Registration. Since the calls will be coming from known peer (IP address of SIP Trunking service q.x.y.z
in our example above) Elastix will accept them without requiring any further authentication.
To configure Elastix server to work with GoTrunk SIP Trunk using SIP Credentials authentication the following changes are required:
Follow steps below to add SIP Trunk:
Trunks
.Add SIP Trunk
link.gotrunk
PEER Details
field (replace eu.st.ssl7.net
with amn.st.ssl7.net
if you want to use North America POP):type=peer host=eu.st.ssl7.net context=from-trunk qualify=yes defaultuser=outbound_sip_username ; <- replace with your Outbound SIP Username remotesecret=outbound_sip_password ; <- replace with your Outbound SIP Password
Register String
:incoming_sip_username:incoming_sip_password@eu.st.ssl7.net
Submit
button.To verify your Elastix server has correctly registered on GoTrunk network follow steps below:
PBX
menu.Tools
.Asterisk-Cli
.sip show registry
Execute
button.Registered
. Any other state indicates communications problem (firewall / NAT issue) between your Elastix server and GoTrunk network or incorrect Register string in your trunk configuration.Next follow "Routing configuration" instructions below.
Follow steps below to add Outbound route:
Outbound Routes
.9_outside
link.gotrunk
from drop down list.Submit Changes
button.For each of the Telephone Numbers on your GoTrunk account follow steps below to add Inbound route:
Inbound Routes
.Add Inbound Route
button.Description
field.DID Number
field.Submit
button.Note: make sure to click Apply Config
button in top right corner of the page to reload your FreePBX configuration.