Re: [asterisk-users] includes in realtime ??
Hello ppl, follow up on a somewot old post. I set rtcachefriends=no and voila! changes to codecs, etc are immediately reflected! now.. that duz raise some issues .. hmmm cheerz Ben. Douglas Garstang wrote: If you want to use MWI, and I imagine most people would, you have to cache your realtime data, which means that changes to the tables do not become effective immediately. They become effective after you prune the entry in memory. Doug. -Original Message- From: RR [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 05, 2006 12:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] includes in realtime ?? Ben, The family name is not sipuser, its sipusers. So try this command realtime load sipusers name username and see if you get nothing. What about? realtime load sipusers username username ? To answer your question, any change in the tables holding this sip users information comes into affect immediately. That's the whole point of realtime :) Cheers, \R ___ --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] includes in realtime ??
lol RR. will def do some RnD on this one, and wil get back. have put this on the back burner for now. thanks again. cheerz Ben RR wrote: I use rtcachefriends=yes and any changes I make in my database become effective immediately along with also getting the MWI functionality. Even though what you say makes sense. Go figure! Ben, yeah if it shows it's loaded then it's there for sure. Sorry I asked for it as in your module listing there wasn't any of these modules. I'm at the end of the rope on troubleshooting your issue. Maybe more detail is needed. Esp when you're saying that your sip.conf general section has just two entries. Where's the rest of it, not that a lot needs to necessarily be there if you're not doing anything too tricky. But I would go with removing the rtcache command from the sip.conf file and try and get realtime working in realtime, if that doesn't sound too whacked, just in case it's working off of some cached data, which is why your old codec selection seems to still work even after you change it. Have you looked in your asterisk log file (full) to see if its complaining about errors when you do a realtime load command? The only time my realtime load comes back empty is when it's got a permission problem of some sort on the DB side and one time it happened because of some delay that was introduced coz of some heavy logging or something, don't quite remember 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 ___ --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] includes in realtime ??
Ben, The family name is not sipuser, its sipusers. So try this command realtime load sipusers name username and see if you get nothing. What about? realtime load sipusers username username ? To answer your question, any change in the tables holding this sip users information comes into affect immediately. That's the whole point of realtime :) Cheers, \R ___ --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] includes in realtime ??
Exactly my point! In my earlier mail, I had a typo in my command. I meant n again tried the command realtime load sipusers name 4000 and also realtime load sipusers username 4000 Its not working yet! Also, if Realtime, I shudn't even be having the need to use the realtime load commands!! I shud change the values in sql, and wham!! it shud be reflected in the call. cheerz, Ben. RR wrote: Ben, The family name is not sipuser, its sipusers. So try this command realtime load sipusers name username and see if you get nothing. What about? realtime load sipusers username username ? To answer your question, any change in the tables holding this sip users information comes into affect immediately. That's the whole point of realtime :) Cheers, \R ___ --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] includes in realtime ??
Ben, that's exactly how it is, the load command is only for you to see what's being pulled from the database and to test if realtime has been configured properly. If you see nothing, then I suspect realtime for you isn't really working and the calls that are working are being looked up in the local conf file. You might have to start doing some toubleshooting. What does your extconfig.conf look like? You might wanna post it here. Also, remove or comment out any extensions related info from sip*.conf files. What's the output if you type: asterisk -rx sip show settings | grep -i realtime on the linux command line? Lastly, ensure there's no errors logged with regards to connectivity to the database. Many pieces need to be in sync for it to work properly. I use it with UnixODBC - FreeTDS - MS SQL Server and it works beautifully :) If you're using a local MySQL database, it should be a piece of cake. Check you're loading the res_mysql module, check for config issues in res_mysql.conf and ensure yur user has permissions to access your asterisk database. Hard to suggest how to do all that without knowing ur exact setup. Sorry, the best I can do for now :) Goodluck \R ___ --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] includes in realtime ??
:) , done all!! neway, lemme know if am overlooking something. extconfig.conf == sipusers = mysql,astDb,sip_conf sippeers = mysql,astDb,sip_conf voicemail = mysql,astDb,voicemail_conf extensions = mysql,astDb,extensions_conf sip.conf has got all entries commented, except for [general] context=default rtcachefriends=yes (hmmm.. is the rtcache the culprit??thats my next investigation! but disabling has got issues with VoiceMail Waiting indication etc) Neway, carrying on... sip show settings === Global Settings: SIP Port: 5060 Bindaddress:0.0.0.0 Videosupport: No AutoCreatePeer: No Allow unknown access: Yes Promsic. redir: No SIP domain support:No Call to non-local dom.:Yes URI user is phone no: No Our auth realmasterisk Realm. auth:No Always auth rejects:No User Agent: Asterisk PBX MWI checking interval: 10 secs Reg. context: (not set) Caller ID: asterisk From: Domain: Record SIP history: Off Call Events:Off IP ToS: 0x0 OSP Support:No SIP realtime: Enabled Global Signalling Settings: --- Codecs: none Relax DTMF: No Compact SIP headers:No RTP Timeout:0 (Disabled) RTP Hold Timeout: 0 (Disabled) MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Pedantic SIP support: No Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Default Settings: - Context:default Nat:RFC3581 DTMF: rfc2833 Qualify:0 Use ClientCode: No Progress inband:Never Language: (Defaults to English) Musicclass: default Voice Mail Extension: asterisk Realtime SIP Settings: -- Realtime Peers: Yes Realtime Users: Yes Cache Friends: Yes Update: Yes Ignore Reg. Expire: No Auto Clear: 120 Modules loaded = *CLI show modules like res Module Description Use Count res_musiconhold.so Music On Hold Resource 1 res_indications.so Indications Configuration0 res_crypto.so Cryptographic Digital Signatures 1 res_adsi.soADSI Resource1 res_odbc.soODBC Resource0 res_config_odbc.so ODBC Configuration 1 res_agi.so Asterisk Gateway Interface (AGI) 0 res_monitor.so Call Monitoring Resource 1 res_features.soCall Features Resource 1 res_config_mysql.soMySQL RealTime Configuration Driver 0 chan_features.so Feature Proxy Channel0 11 modules loaded Anything else... ??? Theres no issue with mysql connection, cuz changes to extensions is reflected back immediately. cheerz Ben. RR wrote: Ben, that's exactly how it is, the load command is only for you to see what's being pulled from the database and to test if realtime has been configured properly. If you see nothing, then I suspect realtime for you isn't really working and the calls that are working are being looked up in the local conf file. You might have to start doing some toubleshooting. What does your extconfig.conf look like? You might wanna post it here. Also, remove or comment out any extensions related info from sip*.conf files. What's the output if you type: asterisk -rx sip show settings | grep -i realtime on the linux command line? Lastly, ensure there's no errors logged with regards to connectivity to the database. Many pieces need to be in sync for it to work properly. I use it with UnixODBC - FreeTDS - MS SQL Server and it works beautifully :) If you're using a local MySQL database, it should be a piece of cake. Check you're loading the res_mysql module, check for config issues in res_mysql.conf and ensure yur user has permissions to access your asterisk database. Hard to suggest how to do all that without knowing ur exact setup. Sorry, the best I can do for now :) Goodluck \R ___ --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] includes in realtime ??
Assuming you have the tables as named int he extconfig.conf as well as the database astDB, how about enabling the module app_realtime.so? Also, if you're using mysql, I don't think you need res_odbc, res_config_odbc. Instead try turning on app_realtime.so and pbx_realtime.so and see how you go :) ___ --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] includes in realtime ??
If it shows in the show modules command, it means, the module is loaded, right? If yes, ^CLIshow modules like app_re Module Description Use Count app_realtime.soRealtime Data Lookup/Rewrite 0 app_readfile.soStores output of file into a variable0 app_record.so Trivial Record Application 0 app_read.soRead Variable Application0 4 modules loaded *CLI show modules like pbx_realtime.so Module Description Use Count pbx_realtime.soRealtime Switch 1 1 modules loaded :| RR wrote: Assuming you have the tables as named int he extconfig.conf as well as the database astDB, how about enabling the module app_realtime.so? Also, if you're using mysql, I don't think you need res_odbc, res_config_odbc. Instead try turning on app_realtime.so and pbx_realtime.so and see how you go :) ___ --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] includes in realtime ??
If you want to use MWI, and I imagine most people would, you have to cache your realtime data, which means that changes to the tables do not become effective immediately. They become effective after you prune the entry in memory. Doug. -Original Message- From: RR [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 05, 2006 12:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] includes in realtime ?? Ben, The family name is not sipuser, its sipusers. So try this command realtime load sipusers name username and see if you get nothing. What about? realtime load sipusers username username ? To answer your question, any change in the tables holding this sip users information comes into affect immediately. That's the whole point of realtime :) Cheers, \R ___ --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] includes in realtime ??
I use rtcachefriends=yes and any changes I make in my database become effective immediately along with also getting the MWI functionality. Even though what you say makes sense. Go figure! Ben, yeah if it shows it's loaded then it's there for sure. Sorry I asked for it as in your module listing there wasn't any of these modules. I'm at the end of the rope on troubleshooting your issue. Maybe more detail is needed. Esp when you're saying that your sip.conf general section has just two entries. Where's the rest of it, not that a lot needs to necessarily be there if you're not doing anything too tricky. But I would go with removing the rtcache command from the sip.conf file and try and get realtime working in realtime, if that doesn't sound too whacked, just in case it's working off of some cached data, which is why your old codec selection seems to still work even after you change it. Have you looked in your asterisk log file (full) to see if its complaining about errors when you do a realtime load command? The only time my realtime load comes back empty is when it's got a permission problem of some sort on the DB side and one time it happened because of some delay that was introduced coz of some heavy logging or something, don't quite remember 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] includes in realtime ??
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Jacob Sent: Monday, September 04, 2006 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] includes in realtime ?? Hello ppl, Is it possible to include contexts in the RealTime scenario?? If not, wots the work around?? Thanks in advance. Ben. ___ --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 Amazing how the wiki has this vast amount of AT LEAST info to start your research on http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions ___ --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] includes in realtime ??
Rushowr wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Jacob Sent: Monday, September 04, 2006 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] includes in realtime ?? Hello ppl, Is it possible to include contexts in the RealTime scenario?? If not, wots the work around?? Thanks in advance. Ben. ___ --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 Amazing how the wiki has this vast amount of AT LEAST info to start your research on http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions ___ --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 Sorry mate. Just slipped the eye. Now to another question, which I tried about. With the Realtime arch, can we change parameters of certain users, say sipusers, at runtime, for e.g. the codec and the change being reflected back immediately? The two SIP users I had, had allow set to gsm;g729;ulaw;alaw, and the two Xlite phones have gsm,ulaw and alaw configured.Calls work fine . I changed the codec(set allow to have only g729). But still the calls go thru. I tried realtime load sipuser name username, to no effect. (anyway, realtime load is only for reading values, if i am not wrong). So is it possible to change user parameters at realtime? or am I missing something again? Thanks again. Ben. ___ --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
