Changes

Jump to: navigation, search

FireBrick SIP Configuration

591 bytes added, 14:57, 11 May 2018
this is important
[[File:2700-small.png|link=:Category:FireBrick]]
 
=Overview=
[[File:Pbvoipicon.png]]Also see: [http://www.firebrick.co.uk/fb2700/voip.php FireBrick VoIP Page] which has more details about the feature, as well as the [http://www.firebrick.co.uk/manuals.php?PRODUCT=2700 FireBrick Manuals] which contain a VoIP section.
 
The FireBrick can be used for VoIP by being a VoIP gateway (FBSIP). Your local (or remote) SIP devices register against the FireBrick, and the FireBrick registers to your SIP provider, in a sense the Firebrick acts as a back-to-back SIP gateway.
We'll set this up so that incoming calls route to a 'ring group', which in turn will route to a number of internal extensions. Outgoing calls from the local phones will all go out via the single SIP account with the service provider.
 
==SIP and NAT==
First, a few comments about SIP, NAT and the FireBrick...
The FireBrick maps ports and IPs for NAT but provides no ALG for SIP or any other protocol. SIP is notoriously difficult with any sort of NAT, with or without ALGs. AAISP will be happy to route a block of IP addresses for use with VoIP phones, and the FireBrick can be configured to use these, and even allocate phones from the same manufacturer the correct IP by DHCP.
Having said this, the a FireBrick acting as a SIP server that itself is not behind NAT will try to cope with SIP clients that are behind NAT. This means that in many cases if there are SIP phones behind NAT than they may well work when registering against the FireBrick.
 
[[File:FBVoIPDia.png|border]]
 
== Prioritising VoIP Traffic==
The FireBrick has built in configuration to give priority to small packets. This would include RTP as well as DNS, SSH and other real time protocols that send small packets.
 
 
== Add a carrier ==
[[File:FireBrick-VoIP-AA2.png|thumb|Carrier Screenshot]]
 
This is the SIP service that you're connecting to:
 
{| border="1" cellpadding="1" cellspacing="1" class="wikitable"
|-
! Config Item
|-
| registrar
| registrarvoiceless.aasipaa.conet.uk
| The SIP Registrar server, supplied by the carrier
|-
| proxy
|
| proxy.aasip.co.uk
| Not normally needed if using AAISP
| The SIP Proxy server, supplied by the carrier
|-
| extn
| 100
| Internal extension number '''where incoming calls are routed to (ege.g. a user or a group)'''
|-
| allow
| 81.187.30.110-119
| IPs that are allowed to talk SIP to us (iei.e. the carriers IPs). (Optional)
|-
| display-name
 
xml:
<source lang=xml>
<carrier name="AASIPAAISP" display-name="Main" allow="81.187.30.110-119" registrar="registrarvoiceless.aasipaa.conet.uk" proxy="proxy.aasip.co.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.
If you have further SIP accounts with carriers then add those too.
 
==Next create some users:==
[[File:FireBrick-VoIP-AA3.png|thumb|VoIP User Screenshot]]
 
These will be your local SIP user accounts that your telephones use to register against the FireBrick with.
Go to: Config - Edit - Setup - Edit VoIP config
Add new VoIP user:
 
{| border="1" cellpadding="1" cellspacing="1" class="wikitable"
|-
! Config Item
| allow
| x.x.x.x/24
| IPs that are allowed to register, put in your LAN ip addresses, ege.g.,
|-
| ddi
| +441234567890
| The full number for this user, iei.e. same as the carrier's number assigned to you.
|-
| extn
 
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="AASIPAAISP" max-calls="1"/>
 
</source>
 
You can repeat this process for your other users, changing the extn each time, ege.g. 102, 103 etc.
At this point you can configure your SIP phones to register to the FireBrick with the credentials you've specified above.
You can then test by calling each other using the extn numbers assigned.
Go to: Config - Edit - Setup - Edit VoIP config
Click Add New
{| border="1" cellpadding="1" cellspacing="1" class="wikitable"
|-
| name
| display-name
| Main
| The name that will display on the phone, ege.g.
|-
| extn
| ring
| 101 102 103
| space separateseparated list of the internal extension numbers to ring
|-
| type
| Ring All
| This is the ring type, ege.g. to ring all at once etc.
|}
Click save.
 
xml:
<source lang=xml>
<group name="Main" display-name="Main" extn="100" ddi="+441234567890" ring="101 102 103" type="all"/>
</source>
 
==Firewall==
SIP and RTP trafficYou will need to beopen allowedthe infirewall if you are actually fire-walling traffic to the FireBrick. This- willoften needpeople tojust befirewall fromtraffic to the carrierLAN, butand alsotherefore fromall externaltraffic SIPto phonesthe ifFireBrick youis have anyallowed.
Take a look at this wiki page for more info: [[FireBrick_2700#VoIP_Rules FireBrick_2700#VoIP_Rules|FireBrick and VoIP Firewall]]
 
SIP and RTP traffic will need to be allowed into the FireBrick. This will need to be from the carrier, but also from external SIP phones if you have any.
Take a look at this wiki page for more info: [[FireBrick_2700#VoIP_Rules FireBrick_2700#VoIP_RulesVoIP Rules|FireBrick and VoIP Firewall]]
 
[[Category:FireBrick VoIP]]
[[Category:VoIP]]
107
edits

Navigation menu