Gabriel. We are using OSPF on our asterisk box. When an interface fails, OSPF switches the default route over to the other interface. :) Fortunately Polycom phones are smart enough to wait for the RTP stream to be re-established. As for OpenSER vs SER... I'm not sure. It really shouldn't make much difference which is used.
-----Original Message-----
From: Gabriel Afana [mailto:[EMAIL PROTECTED]
Sent: Sat 3/11/2006 6:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] Clustering
Doug,
How did get the RTP stream to fail over in progress?
Also, you mentioned your using OpenSER. Why did you choose this
over
the standard SER?
- Gabe
----- Original Message -----
From: "Douglas Garstang" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>; "Asterisk Users Mailing
List -Non-Commercial Discussion" <[email protected]>
Sent: Friday, March 10, 2006 10:49 PM
Subject: RE: [Asterisk-Users] Clustering
> We're doing this. Our Polycom phones point to a domain name that
support
SRV records which gives us a roughly even distribution of calls. We have
OpenSER systems sitting in front of the phones. Each OpenSER system is
configured with different primary/secondary/tertiary Asterisk boxes.
When a
phone registers with SER, it 'copies' the registration down to all the
Asterisk systems.
>
> However, now that I find we allegedly could have used regexten on
Asterisk
to replicate the registrations (yet to see docs on how this works),
that $8k
we spent on systems for OpenSER suddenly seems like money not quite so
well
spent.
>
> Calls to the PSTN are routed from Asterisk back to the OpenSER proxies
where it sends it to the PSTN gateway.
>
> Eventhough it all seems to work quite well, and using OSPF we have
been
able to actually fail an interface on a single OpenSER or Asterisk box
and
fail over an RTP stream (only a few seconds of dead air), due to the
horrible Asterisk documentation, our main challenge has been in
replicating
phone registrations between the Asterisk systems.
>
> It would have been great if the Asterisk product was mature enough to
support Realtime SIP for storing registrations from multiple Asterisk
boxes.
On the surface you'd think it's possible, but every one has a different
opinion about whether it's technically shown to work.
>
> If your going to try and set up a HA Asterisk solution be prepared
for a
really tough time.
>
> Doug.
>
>
>
>
>
>
> -----Original Message-----
> From: Wai Wu [mailto:[EMAIL PROTECTED]
> Sent: Fri 3/10/2006 9:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc:
> Subject: RE: [Asterisk-Users] Clustering
>
>
> If all the sub-servers register themselves to the frontend load
balancer
and support reinvite, the load balancer can decide which server to send
the
call to based on the CPU utilizations of the call processing servers.
I'm
assuming all calls are voip calls here.
>
> -----Original Message-----
> From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ron McCarthy
> Sent: Friday, March 10, 2006 2:22 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Clustering
>
>
> Hello All,
>
> Ive been doing more and more research on trying to setup a
cluster/load
balancer for Asterisk. All the Asterisk boxes would be using a config
that
is the same between them all (via a DB), but we want one location to
point
the phones to, and from there that machine/device will send it to a
Asterisk
server so the call can be processed. I know you cant balance the whole
call,
ie: once the call is started the RTP stream has to go to the same
server,
but a new call could go to a different server if perhaps the 1st server
was
unreachable.
>
> Has anyone tried this, or got this to work? Ive been looking at using
a
Juniper Session Border Controller, but not sure if thats gonna do the
trick,
and then we also have SER..
>
> Any comments would be great!
>
> Thanks
> Ron
>
>
>
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