<div>Hi Zac,</div>
<div> </div>
<div>Thank you so much for your sincere answer. What you brought up is exactly</div>
<div>what I encountered when I tried to find a solution for this, the documentation</div>
<div>is inconsistent and ambiguous, and everywhere I look I end up with outdated </div>
<div>examples that make little or no sense in the good case, or just don't compile </div>
<div>due to being so old in the bad case. This is very frustrating but just by reading </div>
<div>what you wrote was very uplifting for me. </div>
<div> </div>
<div>Thanks again,</div>
<div> </div>
<div>Adi.<br><br> </div>
<div><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 class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">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><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>><br>> _______________________________________________<br>> --Bandwidth and Colocation provided by
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