FireBrick SIP Configuration: Difference between revisions

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[[File:2700-small.png|link=:Category:FireBrick]]

=Overview=

[[File:Pbvoipicon.png]]

~~[[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=2500~~ FireBrick Manuals] which contain a VoIP section.

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Also see: [https://www.firebrick.co.uk/support/knowledge-base/voip/ FireBrick VoIP Page] which has more details about the feature, as well as the [https://www.firebrick.co.uk/support/manuals/0 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.

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==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.

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== 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"

{| border="1" cellpadding="1" cellspacing="1" class="wikitable"

|-

! Config Item

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|-

| registrar

| ~~registrar~~.~~aasip~~.co.uk

| voiceless.aa.net.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 (eg a user or a group)

| Internal extension number '''where incoming calls are routed to (e.g. a user or a group)'''

|-

| allow

| 81.187.30.110-119

| IPs that are allowed to talk SIP to us (ie the carriers IPs). (Optional)

| IPs that are allowed to talk SIP to us (i.e. the carriers IPs). (Optional)

|-

| display-name

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xml:

</syntaxhighlight>

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.

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If you have further SIP accounts with carriers then add those too.

==Next create some users:==

==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"

{| border="1" cellpadding="1" cellspacing="1" class="wikitable"

|-

! Config Item

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| allow

| x.x.x.x/24

| IPs that are allowed to register, put in your LAN ip addresses, eg,

| IPs that are allowed to register, put in your LAN ip addresses, e.g.,

|-

| ddi

| +441234567890

| The full number for this user, ie same as the carrier's number assigned to you.

| The full number for this user, i.e. same as the carrier's number assigned to you.

|-

| extn

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xml:

</syntaxhighlight>

You can repeat this process for your other users, changing the extn each time, eg 102, 103 etc.

You can repeat this process for your other users, changing the extn each time, e.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.

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Go to: Config - Edit - Setup - Edit VoIP config

Click Add New

{| border="1" cellpadding="1" cellspacing="1"

{| border="1" cellpadding="1" cellspacing="1" class="wikitable"

|-

| name

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| display-name

| Main

| The name that will display on the phone, eg

| The name that will display on the phone, e.g.

|-

| extn

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| ring

| 101 102 103

| space ~~separate~~ list of the internal extension numbers to ring

| space separated list of the internal extension numbers to ring

|-

| type

| Ring All

| This is the ring type, eg to ring all at once etc.

| This is the ring type, e.g. to ring all at once etc.

|}

Click save.

xml:

</syntaxhighlight>

==Firewall==

~~SIP and RTP traffic~~ will need to be ~~allowed~~ in to the FireBrick. ~~This~~ ~~will~~ ~~need~~ to be ~~from~~ the ~~carrier~~, ~~but~~ ~~also~~ ~~from~~ ~~external~~ ~~SIP~~ ~~phones~~ if ~~you~~ ~~have any~~.

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.

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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.

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Take a look at this wiki page for more info: [[FireBrick_2700#VoIP_Rules FireBrick_2700#VoIP Rules|FireBrick and VoIP Firewall]]

[[Category:FireBrick]]

[[Category:FireBrick VoIP]]

[[Category:VoIP]]