Jump to content

This is the support site for Andrews & Arnold Ltd, a UK Internet provider. Information on these pages is generally for our customers but may be useful to others, enjoy!

FireBrick SIP Configuration: Difference between revisions

Content deleted Content added
AA-Andrew (talk | contribs)
Reedy (talk | contribs)
Update deprecated tags
 
(10 intermediate revisions by 3 users not shown)
Line 1: Line 1:
[[File:2700-small.png|link=:Category:FireBrick]]

=Overview=
=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=2700 FireBrick Manuals] which contain a VoIP section.

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.
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.
Line 12: Line 16:
==SIP and NAT==
==SIP and NAT==
First, a few comments about SIP, NAT and the FireBrick...
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.
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.


Line 35: Line 40:
This is the SIP service that you're connecting to:
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
! Config Item
Line 67: Line 72:
| extn
| extn
| 100
| 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
| allow
| 81.187.30.110-119
| 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
| display-name
Line 80: Line 85:


xml:
xml:
<syntaxhighlight 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"/>
<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"/>
</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.
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.
Line 87: Line 93:
If you have further SIP accounts with carriers then add those too.
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]]
[[File:FireBrick-VoIP-AA3.png|thumb|VoIP User Screenshot]]


Line 94: Line 100:
Add new VoIP user:
Add new VoIP user:


{| border="1" cellpadding="1" cellspacing="1"
{| border="1" cellpadding="1" cellspacing="1" class="wikitable"
|-
|-
! Config Item
! Config Item
Line 118: Line 124:
| allow
| allow
| x.x.x.x/24
| 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
| ddi
| +441234567890
| +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
| extn
Line 140: Line 146:


xml:
xml:
<syntaxhighlight 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"/>
<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"/>
</syntaxhighlight>


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

You can repeat this process for your other users, changing the extn each time, eg 102, 103 etc.
At this point you can configure your SIP phones to register to the FireBrick with the credentials you've specified above.
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.
You can then test by calling each other using the extn numbers assigned.
Line 155: Line 162:
Go to: Config - Edit - Setup - Edit VoIP config
Go to: Config - Edit - Setup - Edit VoIP config
Click Add New
Click Add New
{| border="1" cellpadding="1" cellspacing="1"
{| border="1" cellpadding="1" cellspacing="1" class="wikitable"
|-
|-
| name
| name
Line 163: Line 170:
| display-name
| display-name
| Main
| Main
| The name that will display on the phone, eg
| The name that will display on the phone, e.g.
|-
|-
| extn
| extn
Line 175: Line 182:
| ring
| ring
| 101 102 103
| 101 102 103
| space separate list of the internal extension numbers to ring
| space separated list of the internal extension numbers to ring
|-
|-
| type
| type
| Ring All
| 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.
Click save.


xml:
xml:
<syntaxhighlight lang=xml>
<group name="Main" display-name="Main" extn="100" ddi="+441234567890" ring="101 102 103" type="all"/>
<group name="Main" display-name="Main" extn="100" ddi="+441234567890" ring="101 102 103" type="all"/>
</syntaxhighlight>


==Firewall==
==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.
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 in to the FireBrick. This will need to be from the carrier, but also from external SIP phones if you have any.
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_Rules|FireBrick and VoIP Firewall]]
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:FireBrick VoIP]]
[[Category:VoIP]]
[[Category:VoIP]]