Hi I have some question about mysql realtime. 1) Anyone know exactly if there is a specific order to declare sip table column for realtime ? In which file can I find that order ? 2) In my extconfig.conf, [settings] are : sipusers => mysql,general,siptable sippeers => mysql,general,siptable so means that I use realtime dynamic exactly ? Is it normal if some parameters from sip.conf still used ? I notice when I change some values in my mysql database, these new values are not directly updated when I reload chan_sip.so module and tape CLI command "sip show peer <peername>". Is it normal ? When I reload chan_sip.so, it seems that connected terminals are no longer detected by Asterisk because when I tape CLI command "sip show peers", there is no results displayed. Any reflexions about that ? thanks in advance for your answers. regards. Harry. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090820/974b2b35/attachment.htm
Hi The column order in your mysql sip table is irrelevant (Example sip table here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip) All generic parameters are still taken from sip.conf and you must set rtcachefriends=yes If you change anything in your mysql sip table you do not need to reload the modue, what you need to do is sip prune realtime <peername> from the CLI As stated previously, you should never have to reload the sip module once realtime is working properly Hope this all helps Ish harry R wrote:> Hi > > I have some question about mysql realtime. > 1) Anyone know exactly if there is a specific order to declare sip > table column for realtime ? In which file can I find that order ? > 2) In my extconfig.conf, [settings] are : > sipusers => mysql,general,siptable > sippeers => mysql,general,siptable > > so means that I use realtime dynamic exactly ? > Is it normal if some parameters from sip.conf still used ? > I notice when I change some values in my mysql database, these new > values are not directly updated when I reload chan_sip.so module and > tape CLI command "sip show peer <peername>". Is it normal ? > When I reload chan_sip.so, it seems that connected terminals are no > longer detected by Asterisk because when I tape CLI command "sip show > peers", there is no results displayed. Any reflexions about that ? > > thanks in advance for your answers. > > regards. > > Harry. > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062
On Thursday 20 August 2009 06:54:12 harry R wrote:> Hi > > I have some question about mysql realtime. > 1) Anyone know exactly if there is a specific order to declare sip table > column for realtime ? In which file can I find that order ?There is no specific order, but columns are processed in the order in which they are found in the table. That means if you use the combination of deny/permit and disallow/allow, then those columns should be found in that order. There is a sample table layout in contrib/scripts/sip-friends.sql.> 2) In my extconfig.conf, [settings] are : > sipusers => mysql,general,siptable > sippeers => mysql,general,siptable > > so means that I use realtime dynamic exactly ? > Is it normal if some parameters from sip.conf still used ?Dynamic realtime is in addition to, not instead of, sip.conf.> I notice when I change some values in my mysql database, these new values > are not directly updated when I reload chan_sip.so module and tape CLI > command "sip show peer <peername>". Is it normal ?I just fixed a bug (yesterday) related to realtime and the use of the 'sip reload' command. That is available in the latest 1.6 branches. The bug did not affect 1.4, however.> When I reload chan_sip.so, it seems that connected terminals are no longer > detected by Asterisk because when I tape CLI command "sip show peers", > there is no results displayed. Any reflexions about that ?They won't be found in the CLI command until Asterisk receives another packet from that peer and a load from the database is forced. -- Tilghman & Teryl with Peter, Cottontail, Midnight, Thumper, & Johnny (bunnies) and Harry, BB, & George (dogs)
"If you change anything in your mysql sip table you do not need to reload the modue, what you need to do is sip prune realtime <peername> from the CLI" Without reload prune does not take effect in 1.4.x And after reload all registrations are lost. So basically Asterisk Realtime is big mess from our experience and is totally unusable. We ended making #exec based script which takes data from DB and forms static configuration on each reload. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ishfaq Malik Sent: 2009 m. rugpj??io 20 d. 15:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mysql sip realtime Hi The column order in your mysql sip table is irrelevant (Example sip table here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip) All generic parameters are still taken from sip.conf and you must set rtcachefriends=yes If you change anything in your mysql sip table you do not need to reload the modue, what you need to do is sip prune realtime <peername> from the CLI As stated previously, you should never have to reload the sip module once realtime is working properly Hope this all helps Ish
I have to disagree with you there, we use 1.4.17 and sip prune realtime works fine Mindaugas Kezys wrote:> "If you change anything in your mysql sip table you do not need to reload > the modue, what you need to do is > sip prune realtime <peername> > from the CLI" > > Without reload prune does not take effect in 1.4.x > > And after reload all registrations are lost. > > So basically Asterisk Realtime is big mess from our experience and is > totally unusable. > > We ended making #exec based script which takes data from DB and forms static > configuration on each reload. > > Regards, > Mindaugas Kezys > http://www.kolmisoft.com > VoIP Billing and Routing Solutions > > > > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ishfaq Malik > Sent: 2009 m. rugpj??io 20 d. 15:56 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] mysql sip realtime > > Hi > > The column order in your mysql sip table is irrelevant > (Example sip table here > http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip) > > All generic parameters are still taken from sip.conf and you must set > rtcachefriends=yes > > If you change anything in your mysql sip table you do not need to reload > the modue, what you need to do is > sip prune realtime <peername> > from the CLI > > As stated previously, you should never have to reload the sip module > once realtime is working properly > > Hope this all helps > > Ish > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062
We use 1.4.18.1 and 1.4.26.1 and it does not work - settings are not changed after prune, asterisk must be reloaded, sip reload or iax2 reload makes changes. But after that all devices loose registration. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ishfaq Malik Sent: 2009 m. rugpj??io 21 d. 12:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mysql sip realtime I have to disagree with you there, we use 1.4.17 and sip prune realtime works fine Mindaugas Kezys wrote:> "If you change anything in your mysql sip table you do not need to reload > the modue, what you need to do is > sip prune realtime <peername> > from the CLI" > > Without reload prune does not take effect in 1.4.x > > And after reload all registrations are lost. > > So basically Asterisk Realtime is big mess from our experience and is > totally unusable. > > We ended making #exec based script which takes data from DB and formsstatic> configuration on each reload. > > Regards, > Mindaugas Kezys > http://www.kolmisoft.com > VoIP Billing and Routing Solutions > > > > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ishfaq Malik > Sent: 2009 m. rugpj??io 20 d. 15:56 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] mysql sip realtime > > Hi > > The column order in your mysql sip table is irrelevant > (Example sip table here > http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip) > > All generic parameters are still taken from sip.conf and you must set > rtcachefriends=yes > > If you change anything in your mysql sip table you do not need to reload > the modue, what you need to do is > sip prune realtime <peername> > from the CLI > > As stated previously, you should never have to reload the sip module > once realtime is working properly > > Hope this all helps > > Ish > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
2009/8/21 Ishfaq Malik <ish at pack-net.co.uk>> I have to disagree with you there, we use 1.4.17 and sip prune realtime > works fine >After a few test, I notice these events when I use asterisk+mysql+realtime+sip 1) after a "sip prune realtime <peername>", <peername> will not be reachable by another terminal except if <peername> make a call or register again. 2) if I set regcontext=<context> [general] setting in sip.conf, extension added automatically in <context> when a peer register and after a "module reload" or a "sip prune realtime <peername>" that extension do not disappear. 3) after a few time, extension added automatically in <context> when a peer register disappear. I dont know how and why, I dont have any msg from asterisk system about that. 4) in my sip table, after a few time, my defaultuser column value for a peer became name column value. (e.g. defaultuser=toto for peer 101 and after a few time defaultuser=101 -_-) have anyone who find an issue to solve these ? could it be caused by my realtime setting in sip.conf ? here a sample of my realtime setting when a do "sip show settings": realtime peers : yes realtime regs : no cache friends : yes update : yes ignore reg expire : no save sys. name : yes auto clear : 120 remind : I use asterisk 1.6, mysql 5 Harry. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090821/3c61fdfe/attachment.htm
Have you set the qualify column in the sip table? harry R wrote:> > > 2009/8/21 Ishfaq Malik <ish at pack-net.co.uk <mailto:ish at pack-net.co.uk>> > > I have to disagree with you there, we use 1.4.17 and sip prune > realtime > works fine > > > After a few test, I notice these events when I use > asterisk+mysql+realtime+sip > 1) after a "sip prune realtime <peername>", <peername> will not be > reachable by another terminal except if <peername> make a call or > register again. > 2) if I set regcontext=<context> [general] setting in sip.conf, > extension added automatically in <context> when a peer register and > after a "module reload" or a "sip prune realtime <peername>" that > extension do not disappear. > 3) after a few time, extension added automatically in <context> when a > peer register disappear. I dont know how and why, I dont have any msg > from asterisk system about that. > 4) in my sip table, after a few time, my defaultuser column value for > a peer became name column value. (e.g. defaultuser=toto for peer 101 > and after a few time defaultuser=101 -_-) > > have anyone who find an issue to solve these ? > could it be caused by my realtime setting in sip.conf ? > here a sample of my realtime setting when a do "sip show settings": > realtime peers : yes > realtime regs : no > cache friends : yes > update : yes > ignore reg expire : no > save sys. name : yes > auto clear : 120 > > remind : I use asterisk 1.6, mysql 5 > > Harry. > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062
>Have you set the qualify column in the sip table?yes and default set to yes -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090821/bbf46830/attachment.htm