Hi all, lets say I've registered at several Sip-Providers. Provider A offers best rates but is often too busy to get a line. Sip Provider B is stable (but more expensive). The asterisk box has a high call volume therefore problems at provider A will get obvious after a few calls stalled. In this case astersik shall switch temporarily to provider B but shall test periodically for selected calls if provider A is available again. I 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!? regards, -- Knud A. M?ller
On Wed, 18 Apr 2007 09:04:22 +0200, Knud M?ller wrote:> I > 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@alphaque.com (0 0) http://www.openmalaysiablog.com/ +==========================----oOO--(_)--OOo----==========================+ | 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 | +=========================================================================+
On Wed, 18 Apr 2007, Knud M?ller said something to this effect:> Hi all, > > lets say I've registered at several Sip-Providers. Provider A offers best > rates but is often too busy to get a line. Sip Provider B is stable (but > more expensive). The asterisk box has a high call volume therefore > problems at provider A will get obvious after a few calls stalled. In > this case astersik shall switch temporarily to provider B but shall test > periodically for selected calls if provider A is available again. I 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!?Best way to do this in my opinion is to deputise this logic to a SIP proxy and have Asterisk trunk all of its calls through that. -- Alex Balashov <sasha@presidium.org>