Hi all,
I installed a Linux-HA-cluster with DRBD and Asterisk 1.4 on it.
Actually it might work quite good, failover etc. works, even if this is
not a 0-downtime solution, because current calls are dropped and the
phones not are reachable until they reregister at the Asterisk.
It might work
on Asterisk DRBD/-HA-Cluster wrong
Actually it might work quite good, failover etc. works, even if this
is not a 0-
downtime solution, because current calls are dropped and the phones
not
are reachable until they reregister at the Asterisk.
Apologize for not directly answering your questions
Thorolf,
In the [general] section of sip.conf, set the 'bindaddr' parameter to
the cluster IP. If Asterisk is only bound to the floating interface, it
will respond only from that source IP.
-- Alex
Thorolf Godawa wrote:
Hi all,
I installed a Linux-HA-cluster with DRBD and Asterisk 1.4
On 4/12/09 9:28 AM, Scott L. Lykens wrote:
Apologize for not directly answering your questions, however, I'm
considering playing with Remus and Xen in the future to deal with high
availability without dropping calls.
See http://dsg.cs.ubc.ca/remus/ for some details.
I have no idea if it
On Dec 3, 2009, at 5:05 PM, Matt Riddell wrote:
On 4/12/09 9:28 AM, Scott L. Lykens wrote:
Apologize for not directly answering your questions, however, I'm
considering playing with Remus and Xen in the future to deal with high
availability without dropping calls.
See
Fred Posner wrote:
If you're using just SIP to SIP, a better option would be a pure sip proxy,
ala Kamailio/SER, etc. They can survive a failover without a drop.
Agreed. Even if using transaction-stateful relay mode, as long as a
dialog is nailed up, sequential in-dialog messages
Hi Alex,
In the [general] section of sip.conf, set the 'bindaddr' parameter to
the cluster IP. If Asterisk is only bound to the floating interface,
yeah, that's it :-)
I have not tested failover right now, but registring of the phones now
works!
Thanks a lot,
--
Chau y hasta luego,
Hi Scott,
Apologize for not directly answering your questions, however, I'm
considering playing with Remus and Xen in the future to deal with
high availability without dropping calls.
thanks a lot for the link, it looks quite interesting.
Unfortunately I think (and this is also my
Thorolf Godawa wrote:
Unfortunately I think (and this is also my experience), a virtualized
Asterisk server will not work on higher load and might loose
UDP-VoIP-pakets what will result in a bad voice quality!
Much has been said of this topic. In general, you are correct; the
effects of
