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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!

VoIP Phones - Asterisk: Difference between revisions

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[[File:Asterisk_logo.png]]
[[File:Asterisk_logo.png]]


*IPv6 Works!
*[[IPv6]] Works!


Asterisk is extremely flexible and covering different uses for it is outside the scope of this example as the setup used here was very basic.
Asterisk is extremely flexible and covering different uses for it is outside the scope of this example as the setup used here was very basic.
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Unfortunately the only way around this seems to be to define a peer for each potential IP address that a call can come from. We only have two live "voiceless" servers at present (and a third test box) but it means that as we expand the service in future Asterisk users would have to keep updating their SIP configs. We therefore created a list of 10 servers (C and onwards are just A records pointing to the A server for now). Sadly this means that we needed to created 20 peers as they will all be dual stack!
Unfortunately the only way around this seems to be to define a peer for each potential IP address that a call can come from. We only have two live "voiceless" servers at present (and a third test box) but it means that as we expand the service in future Asterisk users would have to keep updating their SIP configs. We therefore created a list of 10 servers (C and onwards are just A records pointing to the A server for now). Sadly this means that we needed to created 20 peers as they will all be dual stack!


If your Asterisk box is not dual stack you only need to include the IPv4 '''or''' IPv6 hostnames; whichever you're using.
If your Asterisk box is not dual stack you only need to include the IPv4 '''or''' [[IPv6]] hostnames; whichever you're using.


If you are not registering you will need to set up the username, password and host name for the trunk in the "SIP to your server" settings on the control pages. We do not recommend registering unless you're on a dynamic IP or behind NAT.
If you are not registering you will need to set up the username, password and host name for the trunk in the "SIP to your server" settings on the control pages. We do not recommend registering unless you're on a dynamic IP or behind NAT.
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host=j4.voiceless.aa.net.uk
host=j4.voiceless.aa.net.uk


; IPv6 hostnames
; [[IPv6]] hostnames
[voiceless-11](voiceless-common)
[voiceless-11](voiceless-common)
host=a6.voiceless.aa.net.uk
host=a6.voiceless.aa.net.uk
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[voiceless-out]
[voiceless-out]
type=peer
type=peer
remotesecret=outgoingpass ; Our password to their service
remotesecret=outgoingpass ; Our password to their service *some older asterisk versions require secret, not remotesecret, here*
defaultuser=+441234567890 ; Authentication user for outbound
defaultuser=+441234567890 ; Authentication user for outbound
host=voiceless.aa.net.uk
host=voiceless.aa.net.uk
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host=z4.voiceless.aa.net.uk
host=z4.voiceless.aa.net.uk


; Test server IPv6 hostname
; Test server [[IPv6]] hostname
[voiceless-test6](voiceless-common)
[voiceless-test6](voiceless-common)
host=z6.voiceless.aa.net.uk
host=z6.voiceless.aa.net.uk
Retrieved from "http://support.aa.net.uk/VoIP_Phones_-_Asterisk"