[asterisk-users] SIP failover between Sip Providers
dinesh at alphaque.com
Wed Apr 18 00:21:00 MST 2007
On Wed, 18 Apr 2007 09:04:22 +0200, Knud Müller wrote:
> think it can be done by using the dialplan and the database to store the
> statistical information but maybe there is an easier way that integrates
> better with asterisk!?
i dont think you'd even need a database with statistics. just have all
calls sent to provider A with an automatic failover to provider B if the
call can't be completed through A. you'd need to go look at the DIALSTATUS
variable for that.
Regards, /\_/\ "All dogs go to heaven."
dinesh at alphaque.com (0 0) http://www.openmalaysiablog.com/
| for a in past present future; do |
| for b in clients employers associates relatives neighbours pets; do |
| echo "The opinions here in no way reflect the opinions of my $a $b." |
| done; done |
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