[Asterisk-Users] Fail over?

[Andy Kuo]

in extensions.conf
 exten => _X.,1,Dial(SIP/${EXTEN}@ip.of.provider.1)
exten => _X.,2,Dial(SIP/${EXTEN}@ip.of.provider.2)


 On 11/11/05, John E. Elkin <jeelkin at palrail.com> wrote:
>
> Maybe its already been posted, but i cant find it...
>   I have an asterisk box running agilevoice (Customer signup and
> provisioning system)
>  I have two sip termination providers. One provides did and termination.
> The other provides just my termination. My big question is.
>   If the termination on provider "A" goes out.. i want my asterisk box to
> route calls to provider "B" how do i make this happen automaticly?
>   John
>
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