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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

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=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.
 
[[File:Pbvoipicon.png]]Also see: [httphttps://www.firebrick.co.uk/fb2700support/knowledge-base/voip.php/ FireBrick VoIP Page] which has more details about the feature, as well as the [httphttps://www.firebrick.co.uk/support/manuals.php?PRODUCT=2700/0 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.
==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.
 
 
xml:
<sourcesyntaxhighlight 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"/>
</syntaxhighlight>
</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.
 
xml:
<sourcesyntaxhighlight 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"/>
</syntaxhighlight>
</source>
 
You can repeat this process for your other users, changing the extn each time, e.g. 102, 103 etc.
 
xml:
<sourcesyntaxhighlight lang=xml>
<group name="Main" display-name="Main" extn="100" ddi="+441234567890" ring="101 102 103" type="all"/>
</syntaxhighlight>
</source>
 
==Firewall==
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