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FireBrick SIP Configuration: Difference between revisions
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This is the SIP service that you're connecting to:
{| border="1" cellpadding="1" cellspacing="1" class="wikitable"
|-
! Config Item
| extn
| 100
| Internal extension number '''where incoming calls are routed to (e.g. a user or a group)'''
|-
| allow
xml:
<source lang=xml>
<carrier name="AAISP" display-name="Main" allow="81.187.30.110-119" registrar="voiceless.aa.net.uk"username="01234567890" password="secret" extn="100" comment="Main Office Number"/>
</source>
If you go to Status - VoIP, you should see the Carrier listed with an expiry - this shows that the FireBrick is registered to the server.
Add new VoIP user:
{| border="1" cellpadding="1" cellspacing="1" class="wikitable"
|-
! Config Item
xml:
<source lang=xml>
<telephone name="John" display-name="John" username="john" password="secret" allow="192.168.1.0/24" ddi="+441234567890" extn="101" carrier="AAISP" max-calls="1"/>
</source>
You can repeat this process for your other users, changing the extn each time, e.g. 102, 103 etc.
Go to: Config - Edit - Setup - Edit VoIP config
Click Add New
{| border="1" cellpadding="1" cellspacing="1" class="wikitable"
|-
| name
| ring
| 101 102 103
| space
|-
| type
xml:
<source lang=xml>
<group name="Main" display-name="Main" extn="100" ddi="+441234567890" ring="101 102 103" type="all"/>
</source>
==Firewall==
You will need to open the firewall if you are actually fire-walling traffic to the FireBrick - often people just firewall traffic to the LAN, and therefore all traffic to the FireBrick is allowed.
SIP and RTP traffic will need to be allowed
Take a look at this wiki page for more info: [[FireBrick_2700#VoIP_Rules FireBrick_2700#VoIP Rules|FireBrick and VoIP Firewall]]
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