{{tag>Setup Asterisk FreePBX PBX Configuration}}
====== Getting Started with Asterisk/FreePBX ======
This guide gives a guideline on setting up outbound calling via SureVoIP. It is recommended you use a GUI for setting up Asterisk, such as FreePBX, as it makes setting up a lot easier, and minimises potential for mistakes, which can be very costly if your PBX is compromised.
  * Make sure you have purchased an account with [[http://www.surevoip.co.uk|SureVoIP]]
  * Ensure your firewall has been set up as outlined in [[..:..:NAT and Firewall Settings]]

Please note that this set up guide is for guidance only - it is up to yourself to ensure your phone system has been correctly configured.

SureVoIP can not be held responsible for any damages or losses caused by using this set up guide. 

If you would like for SureVoIP to look over your settings and to help get set up then please [[http://www.surevoip.co.uk/contact-surevoip|get in touch]]. Depending on what is required this may be a chargeable service.
====== Setting Up The Outbound SIP Trunk ======
Log into Asterisk FreePBX Administration

Go to Trunks -> Add Trunk

Trunk Name: **SureVoIP SIP** or something meaningful
Outbound Caller ID: //Your supplied phone number//

Dialled Number Manipulation Rules

^ Prepend ^ Prefix ^ Match Pattern ^ Notes ^
| | 9 | 112 | Emergency Dialling |
| | 9 | 999 | Emergency Dialling |
| | 9 | 101 | Police Non-emergency |
| | 9 | 105 | National power cut and electricity network safety service | 
| | 9 | 111 | NHS Non-emergency |
| | 9 | 119 | COVID-19 Response |
| | 9 | 118XXX | 118 directory enquiries (note: this can be expensive to call) |
| | 9 | 116XXX | Euopean 116 helpline |
| 44 | 90  | ZXXXXXXXX | National Dialling, 10 digits  | 
| 44 | 90 | ZXXXXXXXXX | National Dialling, 11 digits | 
| | 900 | ZXXX. | International Dialling |
| | | 112 | Emergency Dialling |
| | | 999 | Emergency Dialling |
| | | 101 | Police Non-emergency |
| | | 105 | National power cut and electricity network safety service |
| | | 111 | NHS Non-emergency |
| | | 119 | COVID-19 Response |
| | | 118XXX | 118 directory enquiries (note: this can be expensive to call) |
| | | 116XXX | Euopean 116 helpline |
| 44 | 0 | ZXXXXXXXX | National Dialling, 10 digits  | 
| 44 | 0 | ZXXXXXXXXX | National Dialling, 11 digits | 
| | 00 | ZXXX. | International Dialling |

===== Outgoing (PEER) Details =====

Under PEER Details (Outgoing):
<code>
username=xxxxxxxxxxx
type=peer
secret=xxxxxxxxxxxxxxxxx
host=xxx.xxxxxxxx.xx.xx
fromdomain=xxx.xxxxxxxx.xx.xx
fromuser=xxxxxxxxxxx
canreinvite=no
disallow=all
allow=alaw
</code>

Where //xxxxxxxx// is provided in your welcome email. **username** and **fromuser** are the same. **host** is the SureVoIP SIP address. **fromdomain** is the same as **host**.

Note: your PEER Details may vary than that described above, such as the codecs.

===== Incoming (USER) Details =====
This is optional. This information is only required if you prefer not to set //Allow Anonymous Inbound SIP Calls//.

<code>
type=friend
host=x.x.x.x
context=from-trunk
insecure=very
disallow=all
allow=alaw
</code>

You will need to create multiple trunks with the User details.

where //x.x.x.x// is the IP address we supply. [[http://www.surevoip.co.uk/contact-surevoip|Contact us]] for this information.

Be sure to set the //context// relevant to your particular configuration.



===== Registration =====
:!: SureVoIP does not support SIP trunk registration.

Incoming calls to your SIP numbers will go to the SIP URI specified on your account portal.
===== Other Required Settings =====
You will need to go to **Settings -> Asterisk SIP Settings** and set **Allow Anonymous Inbound SIP Calls** to **Yes**.

This is required as incoming calls to your Asterisk system will originate from various servers in the SureVoIP network. Please configure your firewall to only allow incoming VoIP traffic from our IP address ranges. [[http://www.surevoip.co.uk/contact-surevoip|Contact us]] for this info.

Note: if you have configured the USER details (Incoming) settings above then you can leave Allow Anonymous Inbound SIP Calls disabled.

====== Setting up the Inbound Route ======
Go to Inbound Routes -> Add Incoming Route

Give it a meaningful description, such as **SureVoIP Inbound**

DID Number can be left blank or be your provided phone number. If you have multiple phone numbers (DIDs), then put it in here with 01234987654 format (STD with number).

Enter CID Prefix and Music on Hold if required.

Set Destination should be set to where the incoming call should go.

Click Submit

====== Setting up the Outbound Route ======
Go to Outbound Routes -> Add Route

Give it a meaningful name, such as **SureVoIP Outbound**

Dialed Number Manipulation Rules:
^ Prepend ^ Prefix ^ Match Pattern ^ Notes ^
| | | ZXXXXX.| 6 digits or more, first digit 1-9 as validated on outbound route |


Under Trunk Sequence, select the **SureVoIP** Trunk previously created.

Click Submit.

Do not forget to click Apply Configuration.

====== Testing and Troubleshooting ======
If you require technical support, please be sure to provide a **SIP trace** to the technical support team. Guidance on obtaining this can be found at [[troubleshooting:sip_traces|SIP Traces]].

Your router may also need to be configured, and SIP ALG may need to be disabled depending on which router you are using. See [[troubleshooting:sip_alg|SIP ALG]] for guidance on which routers may need adjusting.

Don't forget to configure your firewall correctly - see [[:nat_and_firewall_settings|NAT and Firewall Settings]] for guidance.