It
won't work, unless you make sure that transfers go through the same asterisk
server as the orignal call went through. Using the SER dispatcher won't fix
that.
How do you plan on choosing which
Asterisk server to send the SIP requests? Truly random? Based on some sort of
LCR methodology?
Have you tried using the LCR module for SER to send
the requests to asterisk?
Not sure it would work, but it might be
worth looking at.
N.
On Wed, 27 Sep 2006 21:34:33 +0200, Adi Simon wrote >
Hi Zac, > > Thank you so much for your sincere answer.
What you brought up is exactly > what I encountered when I tried to
find a solution for this, the documentation > is inconsistent and
ambiguous, and everywhere I look I end up with outdated > examples that
make little or no sense in the good case, or just don't compile > due
to being so old in the bad case. This is very frustrating but just by reading
> what you wrote was very uplifting for me. >
> Thanks again, > > Adi. > >
> On 9/27/06, Zac
Amsler <[EMAIL PROTECTED]>
wrote:
Adi,
> > It is possible to do what you are looking for. It is
actually easy. > > There is a problem that I have found with
ser/openser.. Documentation is > difficult to read and some things
are just not there, so you get people > that spend many hours trying
to get these functions to work. In these > days time is money, so the
people that know how to do what you are > seeking.. charge large
amounts of money for a simple 50 line config file. > > I will
tell you that everything you are looking for is documented in >
examples. You will have to piece them together and make them work in
> harmony like the rest of us have. > > I suggest you
look at voip user and piece the config together from > examples
there. It may also help you to read the source code of the > modules
that handle routing in ser. There are a few tricks that are > hidden
in the code. > > I am sorry for my vagueness. I am not able to
share the config > information due to an IP agreement with my
company.(They think it is a > trade secret) > > I wish
you the best. > > Cheers, > Zac Amsler, Network
Operations > Sur-Tel Communications, Inc. & NetIQ Systems, LLC
> * US48, Canada, A-Z Wholesale Termination. > * US48
Origination, Toll Free DIDs. > * Toll Free Termination (FREE).
> > Adi Simon wrote: > > Hi, > > >
> Did anyone actually manage setting up a single SER with multiple
> > Asterisk boxes? > > I particulary have a problem of
keeping the session alive and by that I > > mean directing
> > all the following sip messages to the same asterisk box the
first signal > > was sent (randomally). > > >
> Please don't direct me to Asterisk+At+Large > > <
http://www.voip-info.org/wiki-Asterisk+at+large> or the > >
asterisk_integration > > <http://www.openser.org/dokuwiki/doku.php?id=asterisk_integration
> page > > at openser.org
<http://openser.org> as they are
quite old and useless. > > What I seek are examples of >
> ser.cfg or some advice from someone who actually managed to accomplish
this. > > > > Thanks, > > > > Adi.
> > > > > > > >
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