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<font size="2">How do you plan on choosing which Asterisk server to send the SIP requests? Truly random? Based on some sort of LCR methodology?
<br />
<br />Have you tried using the LCR module for SER to send the requests to asterisk?
<br />
<br />Not sure it would work, but it might be worth looking at.
<br />
<br />N.<b><span style="font-weight: bold;"></span>
<br />
<br />
<br />On Wed, 27 Sep 2006 21:34:33 +0200, Adi Simon wrote</b>
<br />> Hi Zac,
<br />>
<br />> Thank you so much for your sincere answer. What you brought up is exactly
<br />> what I encountered when I tried to find a solution for this, the documentation
<br />> is inconsistent and ambiguous, and everywhere I look I end up with outdated
<br />> examples that make little or no sense in the good case, or just don't compile
<br />> due to being so old in the bad case. This is very frustrating but just by reading
<br />> what you wrote was very uplifting for me.
<br />>
<br />> Thanks again,
<br />>
<br />> Adi.
<br />>
<br />>
<br />> <span class="gmail_quote">On 9/27/06, <b class="gmail_sendername">Zac Amsler</b> <<a href="mailto:list-asterisk@netiqsys.net">list-asterisk@netiqsys.net</a>> wrote:</span>
<blockquote style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;" class="gmail_quote">Adi,
<br />>
<br />> It is possible to do what you are looking for. It is actually easy.
<br />>
<br />> There is a problem that I have found with ser/openser.. Documentation is
<br />> difficult to read and some things are just not there, so you get people
<br />> that spend many hours trying to get these functions to work. In these
<br />> days time is money, so the people that know how to do what you are
<br />>
seeking.. charge large amounts of money for a simple 50 line config file.
<br />>
<br />> I will tell you that everything you are looking for is documented in
<br />> examples. You will have to piece them together and make them work in
<br />> harmony like the rest of us have.
<br />>
<br />> I suggest you look at voip user and piece the config together from
<br />> examples there. It may also help you to read the source code of the
<br />> modules that handle routing in ser. There are a few tricks that are
<br />> hidden in the code.
<br />>
<br />> I am sorry for my vagueness. I am not able to share the config
<br />> information due to an IP agreement with my company.(They think it is a
<br />> trade secret)
<br />>
<br />> I wish you the best.
<br />>
<br />>
Cheers,
<br />> Zac Amsler, Network Operations
<br />> Sur-Tel Communications, Inc. & NetIQ Systems, LLC
<br />> * US48, Canada, A-Z Wholesale Termination.
<br />> * US48 Origination, Toll Free DIDs.
<br />> * Toll Free Termination (FREE).
<br />>
<br />> Adi Simon wrote:
<br />> > Hi,
<br />> >
<br />> > Did anyone actually manage setting up a single SER with multiple
<br />> > Asterisk boxes?
<br />> > I particulary have a problem of keeping the session alive and by that I
<br />>
> mean directing
<br />> > all the following sip messages to the same asterisk box the first signal
<br />> > was sent (randomally).
<br />> >
<br />> > Please don't direct me to Asterisk+At+Large
<br />> > <<a href="http://www.voip-info.org/wiki-Asterisk+at+large">
http://www.voip-info.org/wiki-Asterisk+at+large</a>> or the
<br />> > asterisk_integration
<br />> > <<a href="http://www.openser.org/dokuwiki/doku.php?id=asterisk_integration">http://www.openser.org/dokuwiki/doku.php?id=asterisk_integration
</a>> page
<br />> > at <a href="http://openser.org">openser.org</a> <<a href="http://openser.org">http://openser.org</a>> as they are quite old and useless.
<br />> > What I seek are examples of
<br />> > ser.cfg or some advice from someone who actually managed to accomplish this.
<br />> >
<br />> > Thanks,
<br />> >
<br />> > Adi.
<br />> >
<br />> >
<br />> >
<br />> > ------------------------------------------------------------------------
<br />> >
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<br />> </blockquote>
<br />
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