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 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060927/0b54bcda/attachment.htm
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. > > > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Douglas Garstang
2006-Sep-27 14:26 UTC
[asterisk-users] SER with multiple asterisk deployment
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@arcdiv.com] Sent: Wednesday, September 27, 2006 2:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users-bounces@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@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> http://www.voip-info.org/wiki-Asterisk+at+large> or the > > asterisk_integration > > < http://www.openser.org/dokuwiki/doku.php?id=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. > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users <http://lists.digium.com/mailman/listinfo/asterisk-users> > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060927/fb475067/attachment.htm
Kristian Kielhofner
2006-Sep-27 14:32 UTC
[asterisk-users] SER with multiple asterisk deployment
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. >Adi, The dispatcher module should do what you want to do. Check it out here: http://www.openser.org/docs/modules/1.1.x/dispatcher.html They claim it is stateless but it should be possible to use the AVPs it sets to direct INVITEs, ACKs, and BYEs to the proper Asterisk (or whatever) boxes. However, you can also "load balance" based on source/destination URIs with the lcr module. P.S. - This is really more of an OpenSER/SER question. Did you try those mailing lists? I'd be happy to help you more there :). -- Kristian Kielhofner
Douglas Garstang
2006-Sep-27 22:19 UTC
[asterisk-users] SER with multiple asterisk deployment
If your referring to using AVP operations to peek into the SIP message, and determine state, good luck finding documentation on that! -----Original Message----- From: Jeremy McNamara [mailto:jj@nufone.net] Sent: Wed 9/27/2006 10:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] SER with multiple asterisk deployment 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. ONCE again, design your system correctly and it won't matter which Asterisk box processes your calls - including transfers. No, I won't elaborate, so don't ask. Jeremy McNamara _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 4502 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060927/d0c83f25/attachment.bin
Adi Simon ha scritto:> 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). >record_route() and loose_route() should help you, AFAIK. They don't? Cheers, Simone.
Justin Tunney
2006-Sep-28 19:13 UTC
[asterisk-users] SER with multiple asterisk deployment
Crew, I wrote an SER module called userdispatcher because dispatcher is a static load balancer and therefore worthless. Userdispatcher is really simple at the moment and 300 redirects calls randomly to registered nodes. Therefore it is fault tolerant and balancing based on current active load can be implemented on a higher level. I have not yet used this code in an actual production environment YET but it appears to work. If anyone wants to help test it, feel free. http://www.lobstertech.com/code/userdispatcher/ NOTE: This is not /officially/ released. If you run a blog or something, please don't post details of this yet, keep it in the mailing list for now. I will make a press release when I feel it works. - Justin Tunney On 9/27/06, Adi Simon <adi.simon@gmail.com> 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 or the asterisk_integration page > at 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. > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > >