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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