Re: [asterisk-users] High Availability with Asterisk
Hi all Thanks for an interesting discussion. I've looked at various options for load balancing Asterisk servers and providing fail over support. One thing is not clear to me is: What happens to queues in a load-balancing environment? On our server, we have various queues with up to 20 incoming calls waiting in each, with typically 1-5 queue members. If incoming calls get placed randomly (or according to some heuristic) on different servers, is there any way that Asterisk can handle queue functionality? Our client sip phones can enter or leave queues as they wish, but each sip phone is only registered on one server at a time - so queue members could be registered at different servers in a load balancing environment. Same goes for incoming calls, going to different servers but eventually ending up in the same queue. I'm not sure if queues would ever work in a load balancing scenario, and I haven't found any information on the net to tell me otherwise. Does anybody have any experience/knowledge of if and how it could work? Best regards Binni -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ad...@3a.hu Sent: 8. marts 2014 21:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] High Availability with Asterisk My approach (in theory only, so please correct me if I'm wrong) would be to run asterisk on multiple boxes (one each). A dedicated monitoring box (nagios? custom scripts?) would perform frequent checks against the boxes (one of my previous projects one asterisk was using call files to demonstrate its health to another one). If a box fails, I would simply redirect/reroute its traffic to another one, using network solutions. Such as shutting down the production interface of a suspectedly failed asterisk box, having an idle one pick up its IP address, or using load balancing / routing / NAT to redirect the client's traffic to a standby box. My approach is based on the experience that linux based HA tools are often not free, or don't scale well, or engineered to circumvent an error in a slower manner (eg. booting a second VM takes too much time). However in the network world, there are well known protocols that were designed to take over in a matter of miliseconds. I do understand that this would not provide 'session' data, so failing over to a different box would mean the need to re-register, could cause calls to drop etc. This might be unacceptable for you. As I said in the beginning, I haven't been building such systems, in my experience a dropped call is not that big of a deal, if it happens because the network cuts over to a different box. This could be handled with a pair of frontend load balancers, where the number of asterisk boxes can be transparent. hope this helps adam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hacking attempt, Asterisk 1.4
Hi all We have an Asterisk server thats been running for a few years now without problems. We have IPTables running, as well as fail2ban and have followed all the security recommendations we have found. Every few weeks we get an attack that lasts about a minute or two, resulting in our AGI script being overloaded. What happens is that somebody seems to be trying to connect from our server in my cdrs log I can see that they use a four digit number for source, destination and caller id, e.g. clid: 7321 src: 7321 dst: 7321 channel: SIP/xx.xx.xx.xx- xx.xx.xx.xx is our server IP. When one of our registered users makes a call the channel is SIP/- where is the SIP user ID. So it looks like a SIP phone trying to call itself, using our Asterisk server IP as SIP user name. Within a couple of minutes the attacker seems to go through some 1 attempts, resulting in our AGI script collapsing from the load. My Asterisk full log shows something like: -- Executing [7321@sip:1] Answer(SIP/xx.xx.xx.xx-b0828f20, ) in new stack -- Executing [7321@sip:2] AGI(SIP/ xx.xx.xx.xx -b0828f20, agi:// xx.xx.xx.xx ) in new stack -- Executing [7321@sip:3] Hangup(SIP/ xx.xx.xx.xx -b6130f70, ) in new stack == Spawn extension (sip, 7321, 3) exited non-zero on 'SIP/ xx.xx.xx.xx -b6130f70' cdr_odbc: Query Successful! -- AGI Script agi:// xx.xx.xx.xx completed, returning 0 Our AGI script refuses to call illegal numbers, while our Asterisk dialplan is a bit more accommodating, mostly because I have had problems figuring out the order in which to put the various rules (I might have another look at that!) Does anybody know how to stop this from happening I cant find the attackers IP number in my logs, and these attacks happen infrequently, and are over quickly, so that I havent had an opportunity to run sip debug during an attack, and I dont want to have it running all the time. Best regards Binni Brynjólfur Þorvarðsson IT Consultant Tlf. +45 88321688 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Experience with virtual servers?
Hi All Does anybody have experience with running Asterisk on virtual servers? I have been experimenting with two suppliers and I am not altogether happy with sound quality etc. Is it perhaps foolish to try and install a production Asterisk server on a virtual machine? With dedicated servers being comparatively cheap (although still several times more expensive than virtual servers), perhaps that is the way I should be going? I have heard someone mention Asterisk friendly VPS providers, how can you tell if they are or aren't friendly? We currently have our Asterisk server running on a five year old single AMD CPU 32 bit machine with 512Mb and that works fine. Even the cheapest virtual server vendors offer servers that seem much more powerful but after testing I am not so sure any more! Any info would be very welcome! Regards Binni -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nat=route ???
Hi, I have a question regarding NAT - I have two Asterisk setups, and a couple of softphones on my laptop to test them. In the first Asterisk I've got nat=yes for all SIP phones. The second setup is identical as far as software is concerned, but the server is running on a VPS with one of the larger VPS hosting services. On this second setup I was able to phone out from my XLite softphone but when I tried phoning in nothing happened, basically because the phone was always UNREACHABLE. It would register fine with Asterisk but then disappear. After nosing around for quite a bit I found a suggestion that I try setting nat=route for the SIP phone, and suddenly it worked both ways. Another SIP phone on the same laptop connects to the first Asterisk server setup (which runs on a dedicated box with fixed IP, not in VPS) but I notice that every minute or so Asterisk tells me that the phone is unreachable, then a few seconds later it becomes reachable again. My laptop is currently sitting at home with typical home-Internet configuration (ADSL, Nat, no fixed IP). I did see something somewhere about the big VPS providers using some form of hidden NAT but I don't know what that could mean. My question is: Does this difference in behaviour have something to do with the second server running on a VPS - and are there any drawbacks to using nat=route on all client SIP phones? Best regards Binni Care Solutions Brettesvillesgade 14 9000 Aalborg Telefon: 8832 1600 Mobil: 3020 0868 www.netklinik.dk, www.gangweb.dk, www.caresolutions.dk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stumped trying to find AGI script on my server .....
Hi all I am having some serious trouble finding out how a legacy system is put together. I've almost figured it out, but now I am unable to find what AGI script is actually being used. The system has a lot of code in various states, including Adhearsion and RAGI, but neither seems to be running! Using port for AGI (4573) I've tracked it down to the following: Asterisk machine (X.X.X.X) shows asterisk linking up to port 4573 on another machine (Y.Y.Y.Y): tcp0 0 X.X.X.X:56069 Y.Y.Y.Y:4573ESTABLISHED 29265/asterisk On other machine, Y.Y.Y.Y:4573 shows a pid and process named self: tcp0 0 Y.Y.Y.Y:4573X.X.X.X:56069 ESTABLISHED 9522/self On this same machine, ps aux shows: root 9522 0.2 5.5 120752 115108 ? S 2011 330:43 self I've searched for something that might be named self and have found nothing other than /proc/self which contains my current shell environment. The guys who designed this system are not available to me and they left something of a mess without any documentation. I am in the process of recreating their entire setup (5 boxes) on another site and feel I'm almost done, except for this one small thing of finding out what they use for AGI scripting. I can't mess around with code on the running system to try and trace what is going on so I'm basically stuck. Does anybody have any idea what might be going on? Thanks Med venlig hilsen Binni Care Solutions Brettesvillesgade 14 9000 Aalborg Telefon: 8832 1600 Mobil: 3020 0868 www.netklinik.dk, www.gangweb.dk, www.caresolutions.dk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped trying to find AGI script on my server ..... solved (I think)
Hi, I believe I've found out what this is - one of the previous developers probably left a shell session running Adhearsion in console mode, bless them! -Oprindelig meddelelse- Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Brynjolfur Thorvardsson Sendt: 24. februar 2012 10:34 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [asterisk-users] Stumped trying to find AGI script on my server . Hi all I am having some serious trouble finding out how a legacy system is put together. I've almost figured it out, but now I am unable to find what AGI script is actually being used. The system has a lot of code in various states, including Adhearsion and RAGI, but neither seems to be running! Using port for AGI (4573) I've tracked it down to the following: Asterisk machine (X.X.X.X) shows asterisk linking up to port 4573 on another machine (Y.Y.Y.Y): tcp0 0 X.X.X.X:56069 Y.Y.Y.Y:4573ESTABLISHED 29265/asterisk On other machine, Y.Y.Y.Y:4573 shows a pid and process named self: tcp0 0 Y.Y.Y.Y:4573X.X.X.X:56069 ESTABLISHED 9522/self On this same machine, ps aux shows: root 9522 0.2 5.5 120752 115108 ? S 2011 330:43 self I've searched for something that might be named self and have found nothing other than /proc/self which contains my current shell environment. The guys who designed this system are not available to me and they left something of a mess without any documentation. I am in the process of recreating their entire setup (5 boxes) on another site and feel I'm almost done, except for this one small thing of finding out what they use for AGI scripting. I can't mess around with code on the running system to try and trace what is going on so I'm basically stuck. Does anybody have any idea what might be going on? Thanks Med venlig hilsen Binni Care Solutions Brettesvillesgade 14 9000 Aalborg Telefon: 8832 1600 Mobil: 3020 0868 www.netklinik.dk, www.gangweb.dk, www.caresolutions.dk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?
Hi, I'm working on a small php program for just this. I guess from your question that you have Asterisk writing to a CDR database table, in which case you should be able to use my .php code fairly easily. It's nothing fancy but does give me a graphical presentation of calls/15minute segments. Attached is a screenshot of a graph, I have 1,5+ million entries in the table but there is no noticeable lag in refreshing the graph. At the moment it refreshes only when the button is pressed (the text is in Danish ...) but changing it to refresh automatically every 15 minutes wouldn't be a major problem. I'm working on adding the option of selecting date ranges, it's all still a work in progress! Regards Binni Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af asterisk jobs Sendt: 9. februar 2012 16:36 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR? Hi everyone, I have tons of CDR from an Asterisk with a PRI connection. I want to know som extra details about the calls like the maximum number of calls in peak hours, etc...so I am looking for a php or other type of script that would show this to me in a GUI graphica format. Ideally, it would amazing to feed the asteriskcdrdb table to the program and get back the results without installing anything on the Asterisk server as I don't want to tamper with the server. Is there such a tool? Thanks, attachment: graph.gif-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk vs FreeSwitch vs Asterisk + OpenSIP
I hope I'm not flogging a dead horse here, but the discussion around the whole scalability issue in Asterisk have opened my eyes to a whole lot of issues, making me increasingly confused! We have a fully functioning and stable installation where we offer PBX services to some 15 small firms (basically medical practices). These are based all over the country, with between 2 and 15 SIP phones each. We have a Web front end where each firm can configure their own queues, menus, forwarding etc. My problem is that my bosses want to expand massively, they are currently talking of at least a tenfold increase in the number of clients. I'm fairly certain our Asterisk server won't be able to handle that. Our current 15 clients all have peak usage at the same time (with 2/3 of all traffic between 8 and 9 in the morning). At peak times, we have 20% CPU load with some 100 concurrent calls and a little under one call/second. I have to solve the scalability problem within a relatively short timeframe so starting from scratch with something new is out of the question. My first thought was to add another Asterisk server and use DUNDi load balancing between the two. But looking around and reading the discussion on this list got me to thinking whether some sort of SIP switch or router/proxy could take some load off the Asterisk server(s). One of my main concerns is to change our current setup as little as possible. It's a mishmash of Asterisk, MySQL, Rails and RAGI/RAMI. The original programmers are no longer available to me and I am still very wet behind the ears when it comes to VOIP. So should I be looking at adding e.g. OpenSIP as a sip proxy to our current setup or adding a second (and then a third and a fourth ...) Asterisk server with DUNDi? Or both? Will adding OpenSIP require a change in the way in which we handle SIP peers or require some major reconfiguration of Asterisk? It seems to me that DUNDi requires minimal configuration changes but I don't really know. Any information and recommendations will be greatly appreciated! Regards Binni -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?
Hi, I forgot to add that you are free to use my code, I'll mail it later today. Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Brynjolfur Thorvardsson Sendt: 10. februar 2012 09:47 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR? Hi, I'm working on a small php program for just this. I guess from your question that you have Asterisk writing to a CDR database table, in which case you should be able to use my .php code fairly easily. It's nothing fancy but does give me a graphical presentation of calls/15minute segments. Attached is a screenshot of a graph, I have 1,5+ million entries in the table but there is no noticeable lag in refreshing the graph. At the moment it refreshes only when the button is pressed (the text is in Danish ...) but changing it to refresh automatically every 15 minutes wouldn't be a major problem. I'm working on adding the option of selecting date ranges, it's all still a work in progress! Regards Binni Fra: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af asterisk jobs Sendt: 9. februar 2012 16:36 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR? Hi everyone, I have tons of CDR from an Asterisk with a PRI connection. I want to know som extra details about the calls like the maximum number of calls in peak hours, etc...so I am looking for a php or other type of script that would show this to me in a GUI graphica format. Ideally, it would amazing to feed the asteriskcdrdb table to the program and get back the results without installing anything on the Asterisk server as I don't want to tamper with the server. Is there such a tool? Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk vs FreeSwitch vs Asterisk + OpenSIP
Hi Leandro, that's a really good suggestion. Thanks a lot, I'll certainly give it a try. -Oprindelig meddelelse- Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Leandro Dardini Sendt: 10. februar 2012 14:03 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Asterisk vs FreeSwitch vs Asterisk + OpenSIP 2012/2/10 Brynjolfur Thorvardsson bi...@itanet.nu: I hope I'm not flogging a dead horse here, but the discussion around the whole scalability issue in Asterisk have opened my eyes to a whole lot of issues, making me increasingly confused! We have a fully functioning and stable installation where we offer PBX services to some 15 small firms (basically medical practices). These are based all over the country, with between 2 and 15 SIP phones each. We have a Web front end where each firm can configure their own queues, menus, forwarding etc. My problem is that my bosses want to expand massively, they are currently talking of at least a tenfold increase in the number of clients. I'm fairly certain our Asterisk server won't be able to handle that. Our current 15 clients all have peak usage at the same time (with 2/3 of all traffic between 8 and 9 in the morning). At peak times, we have 20% CPU load with some 100 concurrent calls and a little under one call/second. I have to solve the scalability problem within a relatively short timeframe so starting from scratch with something new is out of the question. My first thought was to add another Asterisk server and use DUNDi load balancing between the two. But looking around and reading the discussion on this list got me to thinking whether some sort of SIP switch or router/proxy could take some load off the Asterisk server(s). One of my main concerns is to change our current setup as little as possible. It's a mishmash of Asterisk, MySQL, Rails and RAGI/RAMI. The original programmers are no longer available to me and I am still very wet behind the ears when it comes to VOIP. So should I be looking at adding e.g. OpenSIP as a sip proxy to our current setup or adding a second (and then a third and a fourth ...) Asterisk server with DUNDi? Or both? Will adding OpenSIP require a change in the way in which we handle SIP peers or require some major reconfiguration of Asterisk? It seems to me that DUNDi requires minimal configuration changes but I don't really know. Any information and recommendations will be greatly appreciated! Regards Binni There are a lots of solutions to asterisk scalability. Each one with its own pros and cons. If you have several small firms, the easiest path will be to duplicate your installation and share your clients among all the servers. Firm01 to Firm15 will be on server01, Firm16 to Firm25 on server02 and so on... However if you have such big numbers of contemporary calls (the max I recorded on one of my server was 60 active calls), maybe you need to think better to high availability, duplicating each server and putting them in high availability. One other way, the one I prefer is to completely share the load among a bunch of servers using mysql multimaster replication and asterisk realtime. Client's phones will use SRV to locate the best server. This way, you can just increase the capacity adding servers and you are completely fault tolerant. Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk V/s FreeSwitch
My Asterisk 1.4.19 happily excepts up to 80 concurrent calls (the most I've seen so far) which sends the CPU load up to ~20% on a fairly old server. In our busiest period, from 8 to 8:05 I see up to 200 incoming calls, somewhat less than one call/second. My superiors want to expand and increase the number of clients significantly and the scalability of Asterisk is beginning to worry me. Someone mentioned a roof of 250 CC in Asterisk after which stability and call quality becomes increasingly affected. My plan is to implement load-balancing using DUNDi with one extra server initially, and a second available on site for further expansion. This should enable me to accommodate ten times our current load without any significant problems (I hope!), and adding more servers is fairly easy (although I guess there are diminishing returns?). When it comes to the long term I must admit I am increasingly looking at trying out FreeSwitch, the configuration might be trickier but scalability is much higher on my list of priorities. Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af virendra bhati Sendt: 7. februar 2012 12:38 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [asterisk-users] Asterisk V/s FreeSwitch Hi List, Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.commailto:virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Headset Options
Hi, Jabra headsets work fine with Polycom. Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Blake Burgess Sendt: 7. februar 2012 05:01 Til: asterisk-users@lists.digium.com Emne: [asterisk-users] Headset Options Hey, I've heard recently from quite a few customers that there's cordless handsets around which don't require a lifter. Is anyone aware of any of these which will work with the cisco 69xx's, 79xx's or any of the current polycom range? -Blake -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk V/s FreeSwitch
According to this article here: http://anders.com/cms/266 the difference mainly lies in how FreeSwitchs handles open channels in comparison with Asterisk. FS uses one thread per channel while * keeps jumping between threads. At least that's how I understand it. Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af virendra bhati Sendt: 8. februar 2012 06:34 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Asterisk V/s FreeSwitch thanks Gilles, After reading these web links. it's pretty clear that FreeSwitch is batter then Asterisk feature, quality wise. But asterisk is easy to used. But the question is still open from my end. How FreeSwitch can support 1000CC but asterisk not ? Because FreeSwitch used XML as configuration and asterisk plan text file ? FreeSwitch used sofia_sip and asterisk used sip ? Asterisk is PBX and FreeSwitch is SoftSwitch ? On Tue, Feb 7, 2012 at 9:10 PM, Gilles codecompl...@free.frmailto:codecompl...@free.fr wrote: On Tue, 7 Feb 2012 17:08:18 +0530, virendra bhati virbh...@gmail.commailto:virbh...@gmail.com wrote: Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... Provided Asterisk, even in release 1.8 or 10, does handle much fewer concurrent calls than Freeswitch, you might find the answer in those articles: How does FreeSWITCH compare to Asterisk? www.freeswitch.org/node/117http://www.freeswitch.org/node/117 Asterisk vs FreeSWITCH www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/http://www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/ Asterisk vs. FreeSWITCH www.anders.com/cms/266http://www.anders.com/cms/266 Open Source VoIP: Asterisk or FreeSwitch? www.zdnet.com/blog/greenfield/open-source-voip-asterisk-or-freeswitch/233http://www.zdnet.com/blog/greenfield/open-source-voip-asterisk-or-freeswitch/233 FreeSwitch vs Asterisk www.dslreports.com/forum/r23246683-FreeSwitch-vs-Asteriskhttp://www.dslreports.com/forum/r23246683-FreeSwitch-vs-Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.commailto:virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any one using VICIDIAL?
Hi all, I'm looking at options for installing/writing PBX software and I came across www.vicidial.orghttp://www.vicidial.org which seems to do almost all I need - and is open source and all. I'd very much like to hear from anyone having experience with VICIDIAL, e.g. using it with different versions of Asterisk (the documentation only mentions * up to 1.6) Best regards Binni ITAnet Kirkestien 20 9230 Svenstrup Telefon: 3020 0868 Email: bi...@itanet.numailto:i...@itanet.nu WWW: http://www.itanet.nuhttp://www.itanet.nu/ [cid:image001.gif@01CCC163.B1ACD9C0] inline: image001.gif-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBC problem - static realtime file not loading SOLVED (properly this time)
Hi Warren According to the book I'm using as well as the documentation on Asterisk 1.8 you should remove the musiconhold.conf file from /etc/asterisk if you want to read it from a database. From what you say it looks as if you can have both. Maybe not with the same classes? You are right about the bug ticket. From what I can gather, the database is defined once for the ODBC driver for each connector (in /etc/odbc.ini). To connect Asterisk to ODBC you need to define a class in res_odbc.conf which points to the ODBC connector. After thinking about this I suppose that what is actually going on is this: 1) In odbc.ini you define one or more ODBC connectors. Each connector has one database. 2) In /etc/asterisk/res_odbc.conf you specify one or more ODBC Asterisk connectors, each pointing to an ODBC connector. 3) In /etc/asterisk/extconfig.conf you put in a line that calls the ODBC driver, the Asterisk connector and optional database. So the syntax of a line should be: filename = driver,Asterisk connector [,table name] In my case, the database user and Asterisk connector were both named asterisk, which confused me into thinking that the extconfig.conf file needed the username. That's not very logical, so I tried changing my Asterisk connector name to [asterisk-odbc] and the line in extconfig.conf to: Musiconhold.conf = odbc,asterisk-odbc,asterisk_files This works fine. In the book, all things are named asterisk - the database user, the database and the Asterisk connector. If I had done the same, everything would have worked fine for me. But since I am working on an RoR-based management interface (just for fun ...) I needed a database with the _development extension, and from there everything went wrong. Anyway i've sent in a bug ticket as you suggested, this may of course be something that has changed since version 1.14 , and anyway, the bug is really in the documentation in the Asterisk Wiki and not in Asterisk as such. Regards Binni Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Warren Selby Sendt: 17. december 2011 21:42 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] ODBC problem - static realtime file not loading On Fri, Dec 16, 2011 at 6:06 AM, Brynjolfur Thorvardsson bi...@itanet.numailto:bi...@itanet.nu wrote: snip After connecting, the asterisk user never sends another SQL statement, at least nothing that shows up in the General log. Asterisk is running as root. I've deleted the musiconhold.conf file from /etc/asterisk I had always thought you still needed the musiconhold.conf file with at least one MOH class defined so that asterisk will load the MOH module. Once it loads the module, then it should read from the database as well. I don't know why this works, but it's the way I've always done it. If this behavior resolves your issue, perhaps a bug ticket is in order on https://issues.asterisk.org/jira/ . -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.comhttp://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SOLVED: ODBC problem - static realtime file not loading
Hi, I solved this and thought I'd share it if anyone is interested. The problem was in the extconfig.conf file, where it says database in all the documentation I could find (as well as in the book itself). Apparently it should be name of database user (which, incidently, is asterisk in both cases in the book I'm using). This is from the book(http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/I_section12_tt1465.html#static_realtime) ; /etc/asterisk/extconfig.conf filename.conf = driver,database[,table] This is from the Asterisk Wiki (https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration): family = realtime driver,db name[,table] So following the guides (which admittedly both refer to version 1.8 - while mine is 1.4) I put in my extconfig.conf the following line: musiconhold.conf = odbc,asterisk_development,asterisk_files where asterisk_development is the name of the MySQL database. That didn't work, but this works!: musiconhold.conf = odbc,asterisk,asterisk_files where asterisk is the name of the database user. Another thing: The general log in MySQL never shows any ODBC commands, only a connect entry made by the ODBC driver. I still haven't found out how to spy on the ODBC commands, but that doesn't worry me at the moment. Well, that took me a whole day to sort out ... Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Brynjolfur Thorvardsson Sendt: 16. december 2011 13:06 Til: asterisk-users@lists.digium.com Emne: [asterisk-users] ODBC problem - static realtime file not loading Hi all I'm trying to configure my Asterisk setup to load the musiconhold.conf file from an ODBC connection to MySQL, working through the example given in the excellent book Asterisk: The Definite Guide. I'm using Asterisk 1.4.19 and MySQL 5.1.58. I've configured the ODBC bit and in my GeneralLog on MySQL I can see the asterisk user connecting and sending a few SQL statements, such as SET SQL_AUTO_IS_NULL = 0. After connecting, the asterisk user never sends another SQL statement, at least nothing that shows up in the General log. Asterisk is running as root. I've deleted the musiconhold.conf file from /etc/asterisk Testing odbc from command line ( there is a difference from what the book says, I need to use sudo for isql to work, presumably since * is running as root) $ odbcinst -q -d [MySQL] $echo select 1 | sudo isql -v asterisk-connector +---+ | Connected!| | | | sql-statement | | help [tablename] | | quit | | | +---+ SQL select 1 +-+ | 1 | +-+ | 1 | +-+ SQLRowCount returns 1 1 rows fetched The extconfig.conf file gets parsed, and looks like this: [settings] musiconhold.conf = odbc,asterisk_development,asterisk_files The modules.conf contains only: [modules] preload = res_odbc.so preload = res_config_odbc.so autoload=yes On starting Asterisk with -cv I get the following: [Dec 16 11:08:38] WARNING[1632]: res_musiconhold.c:1309 load_module: No music on hold classes configured, disabling music on hold. [Dec 16 11:08:38] res_musiconhold.so = (Music On Hold Resource) The second line loads the module in spite of the warning in the first line. The following commands give: *CLI moh show classes *CLI odbc show Name: asterisk DSN: asterisk-connector Pooled: no Connected: yes *CLI module reload res_musiconhold.so *CLImoh show classes *CLI I guess the problem could lie with the database itself but I've checked and double-checked the column names and defs, and the asterisk user has full access rights to the database. The SQL insert for the database looks like this: INSERT INTO `asterisk_files` (`id`, `cat_metric`, `var_metric`, `filename`, `category`, `var_name`, `var_val`, `commented`, `created_at`, `updated_at`) VALUES (1, 1, 1, 'musiconhold.conf', 'default', 'mode', 'files', 0, NULL, NULL), (2, 1, 2, 'musiconhold.conf', 'default', 'directory', '/var/lib/asterisk/moh', 0, NULL, NULL); The two last columns (created_at, updated_at) were created by Rails, but I've also tried pointing Asterisk to a view without those two columns. Basically, the ODBC connection works but it seems as if Asterisk never tries to read the definitions from the database. Any help would be greatly appreciated! Regards Binni ITAnet Kirkestien 20 9230 Svenstrup Telefon: 3020 0868 Email: bi...@itanet.numailto:i...@itanet.nu WWW: http://www.itanet.nuhttp://www.itanet.nu/ [cid:image001.gif@01CCBCA0.81833270] inline: image001.gif-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com
[asterisk-users] ODBC problem - static realtime file not loading
Hi all I'm trying to configure my Asterisk setup to load the musiconhold.conf file from an ODBC connection to MySQL, working through the example given in the excellent book Asterisk: The Definite Guide. I'm using Asterisk 1.4.19 and MySQL 5.1.58. I've configured the ODBC bit and in my GeneralLog on MySQL I can see the asterisk user connecting and sending a few SQL statements, such as SET SQL_AUTO_IS_NULL = 0. After connecting, the asterisk user never sends another SQL statement, at least nothing that shows up in the General log. Asterisk is running as root. I've deleted the musiconhold.conf file from /etc/asterisk Testing odbc from command line ( there is a difference from what the book says, I need to use sudo for isql to work, presumably since * is running as root) $ odbcinst -q -d [MySQL] $echo select 1 | sudo isql -v asterisk-connector +---+ | Connected!| | | | sql-statement | | help [tablename] | | quit | | | +---+ SQL select 1 +-+ | 1 | +-+ | 1 | +-+ SQLRowCount returns 1 1 rows fetched The extconfig.conf file gets parsed, and looks like this: [settings] musiconhold.conf = odbc,asterisk_development,asterisk_files The modules.conf contains only: [modules] preload = res_odbc.so preload = res_config_odbc.so autoload=yes On starting Asterisk with -cv I get the following: [Dec 16 11:08:38] WARNING[1632]: res_musiconhold.c:1309 load_module: No music on hold classes configured, disabling music on hold. [Dec 16 11:08:38] res_musiconhold.so = (Music On Hold Resource) The second line loads the module in spite of the warning in the first line. The following commands give: *CLI moh show classes *CLI odbc show Name: asterisk DSN: asterisk-connector Pooled: no Connected: yes *CLI module reload res_musiconhold.so *CLImoh show classes *CLI I guess the problem could lie with the database itself but I've checked and double-checked the column names and defs, and the asterisk user has full access rights to the database. The SQL insert for the database looks like this: INSERT INTO `asterisk_files` (`id`, `cat_metric`, `var_metric`, `filename`, `category`, `var_name`, `var_val`, `commented`, `created_at`, `updated_at`) VALUES (1, 1, 1, 'musiconhold.conf', 'default', 'mode', 'files', 0, NULL, NULL), (2, 1, 2, 'musiconhold.conf', 'default', 'directory', '/var/lib/asterisk/moh', 0, NULL, NULL); The two last columns (created_at, updated_at) were created by Rails, but I've also tried pointing Asterisk to a view without those two columns. Basically, the ODBC connection works but it seems as if Asterisk never tries to read the definitions from the database. Any help would be greatly appreciated! Regards Binni ITAnet Kirkestien 20 9230 Svenstrup Telefon: 3020 0868 Email: bi...@itanet.numailto:i...@itanet.nu WWW: http://www.itanet.nuhttp://www.itanet.nu/ [cid:image001.gif@01CCBBE7.6E001380] inline: image001.gif-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A few (simple?) questions
Hi all I've been saddled with recreating a running Asterisk PBX setup (with Ruby on Rails). Due to some wrangling between my client and the original developers I am not able to talk to the developers themselves but have been given full SSH access to their servers! My questions are regarding their setup - they have functionality split over several servers as follows (all running CentOS): Server1 MySQL Server2 Ruby on Rails + CSTele Server3 Asterisk 1.4.19 + STUN #1 Server4 Trunk (Asterisk 1.4.19) + STUN #2 Server5 Apache ActiveMQ The system offers PBX services to ~10 small firms and connects via a SIP trunk to a Telecoms company. My questions are as follows: - STUN server - is it necessary (given that there are many free STUN servers on the Internet), and why two? - Why have a separate Asterisk server for the trunk? - Is the Apache Message Queue server necessary? - My info says that server 2 is running CSTele but I have been unable to find a process or program that matches this (except for a comment in a daemon, ast_ami_events.rb, running on Rails server). Can anybody tell me what CSTele might be? Many thanks Binni ITAnet Kirkestien 20 9230 Svenstrup Telefon: 3020 0868 Email: bi...@itanet.numailto:i...@itanet.nu WWW: http://www.itanet.nuhttp://www.itanet.nu/ [cid:image001.gif@01CCBA43.F35D3990] inline: image001.gif-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A few (simple?) questions
Hi, thanks for your answer. I suppose that both the STUN servers and ActiveMQ are there to give a better/more reliable service which is obviously a good idea. From trying to find out some more on the Internet I get the idea that CSTele might have something to do with Circuit Switching. I am guessing that the CSTele server establishes a virtual switching circuit to the queue server and trunk server, possibly through a separate network card (servers 3,4 and 5 all have an extra ethernet card without fixed IP address). Regards Binni -Oprindelig meddelelse- Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Patrick Lists Sendt: 14. december 2011 13:45 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] A few (simple?) questions On 14-12-11 10:18, Brynjolfur Thorvardsson wrote: Hi all I've been saddled with recreating a running Asterisk PBX setup (with Ruby on Rails). Due to some wrangling between my client and the original developers I am not able to talk to the developers themselves but have been given full SSH access to their servers! My questions are regarding their setup - they have functionality split over several servers as follows (all running CentOS): Server1 MySQL Server2 Ruby on Rails + CSTele Server3 Asterisk 1.4.19 + STUN #1 Server4 Trunk (Asterisk 1.4.19) + STUN #2 Server5 Apache ActiveMQ The system offers PBX services to ~10 small firms and connects via a SIP trunk to a Telecoms company. My questions are as follows: -STUN server - is it necessary (given that there are many free STUN servers on the Internet), and why two? Why would you want to rely on a free stun server which can disappear anytime when offering commercial services? I would also deploy my own stun servers for paying customers. -Why have a separate Asterisk server for the trunk? No idea. Maybe the question could be: why have two Asterisk servers? Perhaps for for redundancy/failover? -Is the Apache Message Queue server necessary? No idea. I know BigBlueButton uses Apache MQ Asterisk but I don't know the specifics. -My info says that server 2 is running CSTele but I have been unable to find a process or program that matches this (except for a comment in a daemon, ast_ami_events.rb, running on Rails server). Can anybody tell me what CSTele might be? No idea. Good luck! Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A few (simple?) questions
Hi Carlos and thanks for your answer. To begin with: I am a noob in all telephony/asterisk/ror fields, coming from a Classic ASP/MS background! I've been nosing around in RoR and Asterisk for the last month or so and have managed to create several RoR sites and to get an Asterisk server up and running so me and my boss can phone each other using softphone on a smartphone. So, yes it's going to be fun! And again, thanks for your answer. Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Carlos Alvarez Sendt: 14. december 2011 16:13 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] A few (simple?) questions On Wed, Dec 14, 2011 at 2:18 AM, Brynjolfur Thorvardsson bi...@itanet.numailto:bi...@itanet.nu wrote: I've been saddled with recreating a running Asterisk PBX setup (with Ruby on Rails). Due to some wrangling between my client and the original developers I am not able to talk to the developers themselves but have been given full SSH access to their servers! Jumping in without documentation or help when there is a questionable relationship between the client and developer...this should be a lot of fun. The system offers PBX services to ~10 small firms and connects via a SIP trunk to a Telecoms company. Sounds way over-built, but since we don't know the intent of the architecture nor all the features expected, hard to say. - STUN server - is it necessary (given that there are many free STUN servers on the Internet), and why two? I don't believe so. - Why have a separate Asterisk server for the trunk? Can't think of any reason. - Is the Apache Message Queue server necessary? Necessary is not something that can be answered. In their environment as programmed, probably. In general, can an Asterisk server run without it? Yes. A low-end single x86 server can easily support hundreds of endpoints and dozens of concurrent calls, with all Asterisk services running on a single server. Do you have Asterisk expertise already? RoR, SQL, other telephony...? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A few (simple?) questions
Hi Carlos and thanks for the advice. I agree with you wholeheartedly but I'm not sure if I have much choice in the matter. The system was originally designed to offer PBX services to private clinics and currently handles between 10 and 20, with 70 phone numbers. The guys I work for want to expand into other market segments here in Denmark and my job is to re-install the system on some new servers and start making changes. The code is not very well written, the original developers have totally misunderstood the RVM model in Rails and the Asterix config files are full of unused code and example code. There is also some very sloppy version control in the Rails/Adhearsion files and absolutely no regression testing. But, hey, it seems to work! I would like to start from fresh and re-develop the system, I am not at all confident of being able to just lift the code from the current servers and copy/paste it all onto some new ones and expect it to work. Your solid advice might help me make the case for a fresh start, but whichever way it goes, at least I'll be kept busy ... Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Carlos Alvarez Sendt: 14. december 2011 16:58 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] A few (simple?) questions Getting involved in an existing, and possibly broken system is the wrong way to start with Asterisk. I know, because that's how my career in VoIP started. I had to unlearn a lot of poor practices I learned from that system. But anyway without prior documentation or the ability to get the original design intention, I think your next step is to go right back to the beginning, and gather the user requirements and create a design. Then see if it was solved properly, or you need to start over, or what. Without the basics I don't think you can answer the questions you had. Once you know what was needed and why it was custom-written, you'll probably have all those answers. Just know that in its basic form, to process calls for a normal company, nothing is needed other than one Asterisk server. Everything else is extra, which may or may not be warranted. I've seen a number of deployments that seemed geared more towards making a very profitable complex custom system than just giving the customer the best value. Asterisk is a particularly noob-unfriendly product with a lot of pitfalls and relatively poor documentation. Don't go into it lightly, and always be aware that doing it wrong results in anything from system failures to thousands of dollars in toll fraud costs. On Wed, Dec 14, 2011 at 8:38 AM, Brynjolfur Thorvardsson bi...@itanet.numailto:bi...@itanet.nu wrote: Hi Carlos and thanks for your answer. To begin with: I am a noob in all telephony/asterisk/ror fields, coming from a Classic ASP/MS background! I've been nosing around in RoR and Asterisk for the last month or so and have managed to create several RoR sites and to get an Asterisk server up and running so me and my boss can phone each other using softphone on a smartphone. So, yes it's going to be fun! And again, thanks for your answer. Fra: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] På vegne af Carlos Alvarez Sendt: 14. december 2011 16:13 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] A few (simple?) questions On Wed, Dec 14, 2011 at 2:18 AM, Brynjolfur Thorvardsson bi...@itanet.numailto:bi...@itanet.nu wrote: I've been saddled with recreating a running Asterisk PBX setup (with Ruby on Rails). Due to some wrangling between my client and the original developers I am not able to talk to the developers themselves but have been given full SSH access to their servers! Jumping in without documentation or help when there is a questionable relationship between the client and developer...this should be a lot of fun. The system offers PBX services to ~10 small firms and connects via a SIP trunk to a Telecoms company. Sounds way over-built, but since we don't know the intent of the architecture nor all the features expected, hard to say. - STUN server - is it necessary (given that there are many free STUN servers on the Internet), and why two? I don't believe so. - Why have a separate Asterisk server for the trunk? Can't think of any reason. - Is the Apache Message Queue server necessary? Necessary is not something that can be answered. In their environment as programmed, probably. In general, can an Asterisk server run without it? Yes. A low-end single x86 server can easily support hundreds of endpoints and dozens of concurrent calls, with all Asterisk services running on a single server. Do you have Asterisk expertise already? RoR, SQL, other
