[asterisk-users] SER with multiple asterisk deployment

Wed Sep 27 15:02:33 MST 2006

Yeah... I wasn't really sure. I'm trying to think of a way and nothing comes to mind. The problem is that SER is sort of part stateful and part not, and isn't as concerned with a constant dialog as simply passing the SIP packets effectively.  You might be able to couch some logic somehow that searched for a particular message tag on incoming packets and assigned messages with the same tag an identical flag in the DB (using an AVP), then checked the AVP later to determine the proper direction to route the SIP message. 

It would be easier, I imagine, to write your own SER module to handle the dispatching details and tag searching, though.  

All around, it sounds like it could be a mess. Something to play with, though, if you have time. 

N.

On Wed, 27 Sep 2006 15:25:04 -0600, Douglas Garstang wrote
> 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. 
> -----Original Message-----
> From: sip  [mailto:sip at arcdiv.com]
> Sent: Wednesday, September 27, 2006 2:25  PM
> To: Asterisk Users Mailing List - Non-Commercial  Discussion
> Cc:  asterisk-users-bounces at lists.digium.com
> Subject: Re:  [asterisk-users] SER with multiple asterisk  deployment
> 
> 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 is inconsistent and  ambiguous, and everywhere I look I end up with outdated 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 <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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