Re: [asterisk-users] SER with multiple asterisk deployment
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. ___ --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
Re: [asterisk-users] SER with multiple asterisk deployment
Mainly I have a problem of figuring out how to use them with dispatcher or any other mean of switching between asterisks. Do you have any configuration example of such? On 9/28/06, Simone Ricci [EMAIL PROTECTED] wrote: 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.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: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
Re: [asterisk-users] SER with multiple asterisk deployment
On 9/28/06, Simone Ricci [EMAIL PROTECTED] wrote: 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. ___ --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 On Thu September 28 2006 23:23, Adi Simon [EMAIL PROTECTED] wrote: Mainly I have a problem of figuring out how to use them with dispatcher or any other mean of switching between asterisks. Do you have any configuration example of such? Hi Adi. I highly recommend you move these questions to the SER-users mailing list on iptel.org . The people on that list can most likely answer your questions more easily. Also, the Asterisk-users list is for Asterisk, not SER, discussions. Before you do anything else though, have you searched through the SER-users mailing list? Yours is one of several fairly common questions, and has been addressed in the past. Cheers, -- Nick e: [EMAIL PROTECTED] p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --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
Re: [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 [EMAIL PROTECTED] 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 ___ --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
Re: [asterisk-users] SER with multiple asterisk deployment
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 ___ --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
Re: [asterisk-users] SER with multiple asterisk deployment
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 wrotewas very uplifting for me. Thanks again, Adi. On 9/27/06, Zac Amsler [EMAIL PROTECTED] 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 peoplethat spend many hours trying to get these functions to work. In thesedays 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 inexamples. 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 fromexamples there. It may also help you to read the source code of themodules 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 configinformation due to an IP agreement with my company.(They think it is atrade secret)I wish you the best. Cheers,Zac Amsler, Network OperationsSur-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 ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: 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
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 [EMAIL PROTECTED] 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. ___ --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 ___ --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 ___ --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
RE: [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:[EMAIL PROTECTED]Sent: Wednesday, September 27, 2006 2:25 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionCc: [EMAIL PROTECTED]Subject: Re: [asterisk-users] SER with multiple asterisk deploymentHow 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 wrotewas very uplifting for me. Thanks again,Adi. On 9/27/06, Zac Amsler [EMAIL PROTECTED] 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. ___ --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 ___ --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 ___ --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
Re: [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 ___ --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
RE: [asterisk-users] SER with multiple asterisk deployment
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:[EMAIL PROTECTED] Sent: Wednesday, September 27, 2006 2:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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. ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit
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
RE: [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:[EMAIL PROTECTED] 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 winmail.dat___ --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
Re: [asterisk-users] SER with multiple asterisk deployment
Douglas Garstang wrote: If your referring to using AVP operations to peek into the SIP message, and determine state, good luck finding documentation on that! From: Jeremy McNamara [mailto:[EMAIL PROTECTED] 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. That's funny, Doug, you giving advice to Jeremy. My favorite CS teacher in college, with almost maddening frequency, would answer our questions about the operational characteristics of the software we were working on by tugging ponderously on his chin, looking up at the ceiling, then busting a big grin and then saying, I don't know. What does the source code say? B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
