[asterisk-users] SER with multiple asterisk deployment

[Adi Simon]

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is inconsistent and ambiguous, and everywhere I look I end up with outdated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 <list-asterisk at netiqsys.net> 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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