asterisk users - Dec 2004 - pseudo load balancing?

In an environment with multiple asterisk boxes, each with a 4PRI card
and 4PRIs (92 Zap ports) and oversubscription on SIP peers, is there a
way to get the asterisk box to use the Zap interfaces on another box in
times of congestion? While the oversubscription ration would be
optimized for the number of Zap ports, there is always the possibility
of an "unusual" load.

What about if a single DID number for a single user were configured to
come in on any of the PRIs connected to any of the * boxes, and it ended
up coming in to a different box than the SIP peer is configured for, can
* route the call?

Is 4PRIs too much for a single box (3ghz single or dual cpu) assuming
all calls are either Zap<--->SIP SIP<---->Zap, no low bit rate
codecs
are used (g.711 only), and every SIP peer also has a voicemail box? 

Can this be done? Are there any consultants hanging out here that have
real world experience with this?

Damon

On Thu, 2004-12-09 at 08:19 -0700, Damon Estep wrote:
> In an environment with multiple asterisk boxes, each with a 4PRI card
> and 4PRIs (92 Zap ports) and oversubscription on SIP peers, is there a
> way to get the asterisk box to use the Zap interfaces on another box in
> times of congestion? While the oversubscription ration would be
> optimized for the number of Zap ports, there is always the possibility
> of an "unusual" load.
> 
> What about if a single DID number for a single user were configured to
> come in on any of the PRIs connected to any of the * boxes, and it ended
> up coming in to a different box than the SIP peer is configured for, can
> * route the call?
I think this is the textbook case for needing SER. All your SIP users go
to the SER app and it deligates calls to an asterisk box for PSTN
completion as well as any asterisk box can go to SER and get connected
to a specific SIP user.

> Is 4PRIs too much for a single box (3ghz single or dual cpu) assuming
> all calls are either Zap<--->SIP SIP<---->Zap, no low bit rate
codecs
> are used (g.711 only), and every SIP peer also has a voicemail box? 
It somewhat depends on volume of call setup and teardowns. If your calls
are of a decent length, setup/teardown will be sparse enough to not
cause too much trouble.
-- 
Steven Critchfield <critch@basesys.com>

> > In an environment with multiple asterisk boxes, each with a 4PRI
card
> > and 4PRIs (92 Zap ports) and oversubscription on SIP peers, is there
a
> > way to get the asterisk box to use the Zap interfaces on another box
in
> > times of congestion? While the oversubscription ration would be
> > optimized for the number of Zap ports, there is always the
possibility
> > of an "unusual" load.
> >
> > What about if a single DID number for a single user were configured
to
> > come in on any of the PRIs connected to any of the * boxes, and it
ended
> > up coming in to a different box than the SIP peer is configured for,
can
> > * route the call?
> 
> I think this is the textbook case for needing SER. All your SIP users
go
> to the SER app and it deligates calls to an asterisk box for PSTN
> completion as well as any asterisk box can go to SER and get connected
> to a specific SIP user.
So * still does the zap-sip conversion, but all sip peers register with
SER? Incoming Zap call goes back out to SER than peer, or SER handles a
re-invite to *?

Can SER equally distribute Zap calls among * boxes? Will it know if the
asterisk box is congested on the Zap cannels?

Have you implemented this before? (I would need some paid help)