Sip2sim Source IPs: Difference between revisions
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Customers using asterisk will know that the config for these can be somewhat complex, listing 10 hostnames each for IPv4 and [[IPv6]] as the way asterisk works is to look up an IP for a hostname, pick the first, and check that against the request IP address. Asterisk really needs fixing.
At the moment AAISP have 2 'voiceless' SIP servers.
The "voiceless" call servers use two addresses per server for each of IPv4 and [[IPv6]]. These shall be:
81.187.30.111
81.187.30.113
2001:8b0:0:30::5060::1
2001:8b0:0:30::5060::3
(for the A server)
81.187.30.112
81.187.30.114
2001:8b0:0:30::5060::2
2001:8b0:0:30::5060::4
(for the B server)
You do not need to know if A or B. The additional two addresses will be used as "source" addresses for any request from these servers to you that need authentication. This allows asterisk to be configured separately for authenticated and unauthenticated requests.
Requests may also come from other addresses within the published block when test servers are used, etc.
We may adjust which IPs are which server at a future date as we also have to consider how we expand beyond the two servers in use currently. These IPs will be accessible via DNS as:
a.voiceless.aa.net.uk
a.auth.voiceless.aa.net.uk
b.voiceless.aa.net.uk
b.auth.voiceless.aa.net.uk
...and so on.
If your existing asterisk config is working as per the recommendations, no changes will be needed. The wiki will be updated to explain how you can use these changes.
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