Hello All,
I've been investigating and playing with asterisk to see how it would
work out as a small-medium business pbx to handle mostly
interoffice/branch communication and a possibly communication out to
pstn in later stages of implementation. (All communication would be VoIP
internally with possibly 4 pstn lines at the HQ with either a TDM04B or
spa3000s and 1-2 pstn lines across 7-8 branches with spa3000s).
Still evaluating my options about which devices will be best.
One thing I've been trying to figure out with little luck is regarding
true failover/redundancy design and would like some suggestions from the
list. I've looked through
http://www.voip-info.org/tiki-index.php?page=Asterisk+High+Availability+
Solutions and I can understand the high availability solutions. Though
there seems to be several possible solutions for failover but none
really seems to be the "recommended" way to implement. Most seem just
another "possible" way.
Since my proposed setup is fairly small 20-40 voip phones max maybe
12-20 pots lines my main goal is:
- 2 asterisk boxes, Hopefully possible to be in different sites
- If box1 fails calls will stay up (except those that are pots
connected off that box)
- Rest of the VoIP phones will re-register with box 2. (I
noticed the cisco 79xx have backup proxy so this could be handled by the
phones).
Or is there some way to for asterisk box 1 to pass registrations to box
2 as well?
What about the spa3ks how would they handle re-registration to a
different ip?
- Configs / Voicemails mirrored across both servers (probably
easy done with rsync)
I took a brief look at proxy with SER but I believe there are a few
things I possibly don't understand.
- How does SER determine if an asterisk box is down? Or it SER only for
load balancing?
- Looks like I would need another box or 2 specifically to do SER and
then well that seems to become a point of failure?
What are some ways people implement real failover with asterisk? Or if
there are some other resource that I should look at? Please feel to
point out if I'm way off the mark in anything or my expectations.
Thanks for your time
John Cianfarani
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050627/5614cc32/attachment.htm
