Re: [Asterisk-Users] MySQL and Asterisk
Dan Journo wrote: Is there a guide anywhere which runs through how to set up asterisk with mysql? I've looked and almost all the document misses out relevant information. Thanks Dan Journo What do you want to do with mysql? Did you read on the wiki? There is tons of info there. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk realtime beta
So, you admit that you can do what you want using RealTime Static, but you are just unwilling to do so. So, how is that a limitation if you 'can' do it? -Matthew From: Urban [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 19 Sep 2005 11:03:09 +0200 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk realtime beta Matthew Boehm wrote: I currently not use it due to some limitations in * realtime . Such as? -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My current configuration uses a lot of include statements to split up the context's such as security contexts included per extension (allow national, internation calls etc). Since realtime does not have this type of feature (if you not using static) I decided it was to much work to redeisgn the dialplan at the moment. /urban ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Hardware Interrupts; Who is it?
I've been trying to diagnose why my server has a constant idle time of 90% even when nothing is running. After finally discovering what hi means in 'top' (it means hardware interrupts) I find that this percentage always averages around 7-10%. How can I find out what is causing this constant load of interrupts? I have a Dell 1850 3.0Ghz with on board RAID and 2GB RAM. Anyone else experiencing this? -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] Asterisk realtime beta
That is not a limitation of the Asterisk RealTime Architecture. That is a limitation of libiodbc. -Matthew From: Olivier Taylor [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 19 Sep 2005 18:09:38 +0200 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Subject: RE : [Asterisk-Users] Asterisk realtime beta The limitation is that it doesn't work on freebsd, probably due to libiodbc... That's a limitation, isn't it? Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Matthew Boehm Envoyé : lundi 19 septembre 2005 16:42 À : Asterisk Users Objet : Re: [Asterisk-Users] Asterisk realtime beta So, you admit that you can do what you want using RealTime Static, but you are just unwilling to do so. So, how is that a limitation if you 'can' do it? -Matthew From: Urban [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 19 Sep 2005 11:03:09 +0200 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk realtime beta Matthew Boehm wrote: I currently not use it due to some limitations in * realtime . Such as? -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My current configuration uses a lot of include statements to split up the context's such as security contexts included per extension (allow national, internation calls etc). Since realtime does not have this type of feature (if you not using static) I decided it was to much work to redeisgn the dialplan at the moment. /urban ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?
OK. Perhaps I was not clear. Please read my original post again: I've been trying to diagnose why my server has a constant idle time of 90% even when nothing is running. I did not say 90% usage I said 90% idle. Meaning I have a constant CPU usage of 10%. This 10% measurement comes from the hardware interrupt (hi) from within top: Cpu(s): 3.8% us, 2.1% sy, 0.0% ni, 85.5% id, 0.2% wa, 8.0% hi, 0.4% si Even when all other percentages are at 0%, hi remains around 10%. How can I figure out what is causing all these interrupts? -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?
CPU0 CPU1 0: 85 1703809954IO-APIC-edge timer 8: 0 0IO-APIC-edge rtc 9: 0 1 IO-APIC-level acpi 14: 0 31IO-APIC-edge ide0 177: 0 17840313 IO-APIC-level megaraid 185: 0 1817423967 IO-APIC-level eth0 193: 0 40198530 IO-APIC-level eth1 201: 0 3507106255 IO-APIC-level wanpipe1, wanpipe2, wanpipe3, wanpipe4 NMI: 0 0 LOC: 1633394197 1633394188 ERR: 0 MIS: 0 Any idea what LOC and timer are? I did watch -n1 cat /proc/interrupts for about a minute. The largest movers were timer, LOC and eth0. The others either never change or hardly changed. -Matthew From: Sig Lange [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 19 Sep 2005 15:34:26 -0400 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it? It means you have a piece of hardware that is generating a lot of interupts. Try a few commands like this: # vmstat 1 # watch -n1 cat /proc/interrupts Go through lspci -vb and disable hardware that's not being used. Watch the numbers as they increase. Also check for ERR and MIS On 9/19/05, Rich Adamson [EMAIL PROTECTED] wrote: OK. Perhaps I was not clear. Please read my original post again: I've been trying to diagnose why my server has a constant idle time of 90% even when nothing is running. I did not say 90% usage I said 90% idle. Meaning I have a constant CPU usage of 10%. This 10% measurement comes from the hardware interrupt (hi) from within top: Cpu(s): 3.8% us, 2.1% sy, 0.0% ni, 85.5% id, 0.2% wa, 8.0% hi, 0.4% si Even when all other percentages are at 0%, hi remains around 10%. How can I figure out what is causing all these interrupts? I don't have a clue other then to unload drivers, etc. ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com-- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sig Lange http://www.signuts.net/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk realtime beta
I currently not use it due to some limitations in * realtime . Such as? -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pass through of T.38
Carlos Alperin wrote: I don't get this? Is included on 1.0.9 or is not? I know that a lot of people was trying it, but just to be clear, is T.38 passthrough included on 1.0.9? Thanks, Carlos Alperin No. We are not even sure at this point if T38 will make it into 1.2. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_addon_mysql.so pb
Run the command cdr mysql status from asterisk CLI. What does that say? If it says command not found then the module is not loaded. -Matthew From: alexandre zhang [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 12 Sep 2005 02:11:05 +0800 (CST) To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] cdr_addon_mysql.so pb hi I load cdr_addon_mysql.so without error configuration of cdr_mysql.conf [general] dbhost = localhost dbname = recharge dbuser = root dbpass = ast dbport = 3306 dbsock = /var/lib/mysql/mysql.sock But, I get nothing in the table of cdr of my database. Somebody have an idea ? Thanks you for your help best regards - DO YOU YAHOO!? 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: 回复： Re: [Asterisk-Users] cdr_addon_mysql.so pb
cytrex2*CLI cdr mysql status Connected to [EMAIL PROTECTED], port 3306 using table cdr for 3 days, 18 hours, 4 minutes, 3 seconds. Wrote 62028 records since last restart. That is what you should see. -Matthew From: alexandre zhang [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 12 Sep 2005 04:07:28 +0800 (CST) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: 回复： Re: [Asterisk-Users] cdr_addon_mysql.so pb thanks for ur help Run the command cdr mysql status I got the following msg ' No such command 'cdr mysql' (type 'help' for help)' But, I run ' show modules' cdr_addon_mysql.so is in the list Do u have an idea about it ? Thanks Matthew Boehm [EMAIL PROTECTED] 写道： Run the command cdr mysql status from asterisk CLI. What does that say? If it says command not found then the module is not loaded. -Matthew From: alexandre zhang Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Date: Mon, 12 Sep 2005 02:11:05 +0800 (CST) To: Subject: [Asterisk-Users] cdr_addon_mysql.so pb hi I load cdr_addon_mysql.so without error configuration of cdr_mysql.conf [general] dbhost = localhost dbname = recharge dbuser = root dbpass = ast dbport = 3306 dbsock = /var/lib/mysql/mysql.sock But, I get nothing in the table of cdr of my database. Somebody have an idea ? Thanks you for your help best regards - DO YOU YAHOO!? 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - DO YOU YAHOO!? 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pass through of T.38
Roger Schreiter wrote: Hi, I found some contradicting infos about pass through of T.38 data. I almost had to change my pants when I saw a CVS update this morning adding T38 frame recognition to asterisk. I kept looking for the code that complimented this but haven't seen it yet. And there was no bug reference so I can't help test. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MAX PRI for single server
[EMAIL PROTECTED] wrote: Ah - so the difference between your setup and mine is that you are using Sangoma (presumably) and I'm using Digium. Looks like the Digium is significantly more efficient then. It could also be that I'm using Net-SNMP to query my cpu usage and even when the machine is idle, SNMP reports about 20% CPU usage which is incorrect. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom 501 Multiple Line Instances
I tried following the Wiki page regarding the Polycom 501 and having the same extension appear on all 3 line buttons (just like my cisco) but I'm having no luck. Has anyone else had success in doing this? Perhaps someone who has been successful can update the wiki? Thanks, Matthew http://www.voip-info.org/tiki-index.php?page=Polycom+Soundpoint+IP+501 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Insert Subject Here
Flobi wrote: I've been messing with it for a couple weeks with MySQL. It seems pretty good to me though I have had a couple crashes. I cane' say for sure that the crashes were directly related to RealTime though. Also, I'm still using CVS HEAD 2005-09-06 which was right before the beta release, I think. Flobi, please use the subject line. Its there for a reason. Secondly, if your system is crashing, how do you expect us to help debug the problem if you don't provide any info? Like backtrace's etc.. Read doc/README.backtrace for more info. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Instances of Asterisk (no contexts)
Geoff Karl wrote: I know I have seen something on the mailing list describing how to run more than one instance of Asterisk. I can't find it anymore. What are the things to look for when running more than one copy. Yes, I know about contexts. thanks, Geoff This begs a repeated question: Why? The entire 'point' of contexts is so you don't have to run multiple instances of asterisk. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: MAX PRI for single server
Wayne Gemmell wrote: On Thursday 08 September 2005 16:26, Simone Cittadini wrote: My boss is just asking me if it is possible to stuck 4* TE411P in a Doesn't that equal 16 lines, not 480 lines? Or did I miss something? Yes, you missed something: 4 PRIs = 92 Lines per Card * 4 Cards = 368 Lines That is assuming you have 1 D-chan per span. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MAX PRI for single server
Jason Becker wrote: Sage advice, but out of curiousity what happened to Digium's T3 card (the DS3000P)? IIRC, Digium's T3 card isn't expected to be channelized. Also, IIRC it will have no on-board EC and no on-board encoding so I can't imagine the machine you would need to process that many calls. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MAX PRI for single server
Jason Becker wrote: Hmm, looks like someone in the know needs to update the wiki: http://www.voip-info.org/tiki-index.php?page=Digium+DS3000P Wow. Guess I'm not. I've got a 4 port PRI card in this brand new Dell 1850 3.0Ghz Xeon with 2GB RAM and I run an average of 50-60% CPU usage with just 47 calls. All G711. I guess a 64 bit chip is really that much better eh? -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MAX PRI for single server
Carlos Antunes wrote: Have you seen this? http://www.digium.com/index.php?menu=compatibility Yes, but I'm not using a Digium card. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extensions - Realtime
CVS HEAD/Asterisk 1.2: Is there a way to have the entire extensions.conffile coming from the realtime? Yes. Go read the wiki on RealTime Static. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extensions - Realtime
I don't see your swich statement anywhere. You must define a context [default] then add in the correct switch= statement. -Matthew From: Flobi [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 7 Sep 2005 12:18:26 -0400 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Extensions - Realtime Okay, this doesn't seem to be working. I've gone and deleted my ael file also. I do know my MySQL is set up cause I have my sip, iax and voicemail going through it too. here's the line in extconfig.conf: [settings] extensions.conf = mysql,asterisk,pbx_realtime_extensions in pbx_realtime_extensions, my db table: id name context exten priority app appdata 1 default default _. 1 NoOp Testing CLI show dialplan [ Context 'parkedcalls' created by 'res_features' ] '700' = 1. Park() [res_features] -= 1 extensions (1 priorities) in 1 contexts. =- And when I try to call, I get: Sep 7 12:16:30 NOTICE[14330]: chan_iax2.c:7052 socket_read: Rejected connect attempt from 206.180.254.97, request '[EMAIL PROTECTED]' does not exist Also, this message keeps popping up even when calls aren't going through: Sep 7 12:15:12 NOTICE[14330]: pbx.c:1688 pbx_extension_helper: Cannot find extension context 'default' On 9/7/05, Flobi [EMAIL PROTECTED] wrote: It states that the conf file overrides the static db info, but what about the ael file? Does that override also? BTW, RealTime Static...talk about oxymoron :-) Gotta love it! Flobi On 9/7/05, Matthew Boehm [EMAIL PROTECTED] wrote: CVS HEAD/Asterisk 1.2: Is there a way to have the entire extensions.conffile coming from the realtime? Yes. Go read the wiki on RealTime Static. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Automated Signature: This message is from Flobi of Flobi.com. Visit my website if you like: http://www.flobi.com/ Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- -- Automated Signature: This message is from Flobi of Flobi.com. Visit my website if you like: http://www.flobi.com/ Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extensions - Realtime
The wiki doc's are correct. You are trying to combine two different methods of pulling RealTime extensions and that is why it isn't working as you are expecting. Pick 1 method and all will be revealed. Both are very simple to do. -Matthew From: Flobi [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 7 Sep 2005 13:00:26 -0400 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Extensions - Realtime Nevermind, I figured out that the table is used way differently when doing static. Here's my fixed table. I'll try to explain this in the voip-info doc. id cat_metric var_metric commented filename category var_name var_val 1 0 0 0 extensions.conf default exten _.,1,NoOp(Testing) On 9/7/05, Flobi [EMAIL PROTECTED] wrote: Okay, after noticing an error on this mysql statement after i switched to odbc: SELECT * FROM pbx_realtime_extensions WHERE filename='extensions.conf' and commented=0 ORDER BY filename,cat_metric desc,var_metric asc,category,var_name,var_val,id I added those fields and reloaded...* immediately crashed. I restarted. Now, I'm getting this: *CLI show dialplan [ Context 'NoOp' created by 'pbx_config' ] [ Context 'parkedcalls' created by 'res_features' ] '700' = 1. Park() [res_features] -= 1 extensions (1 priorities) in 2 contexts. =- out of this table: id name context exten priority app appdata filename commented cat_metric var_metric category var_name var_val 1 default default _. 1 NoOp Testing. extensions.conf 0 NULL NULL NULL NULL NULL On 9/7/05, Flobi [EMAIL PROTECTED] wrote: Okay, this doesn't seem to be working. I've gone and deleted my ael file also. I do know my MySQL is set up cause I have my sip, iax and voicemail going through it too. here's the line in extconfig.conf: [settings] extensions.conf = mysql,asterisk,pbx_realtime_extensions in pbx_realtime_extensions, my db table: id name context exten priority app appdata 1 default default _. 1 NoOp Testing CLI show dialplan [ Context 'parkedcalls' created by 'res_features' ] '700' = 1. Park() [res_features] -= 1 extensions (1 priorities) in 1 contexts. =- And when I try to call, I get: Sep 7 12:16:30 NOTICE[14330]: chan_iax2.c:7052 socket_read: Rejected connect attempt from 206.180.254.97, request '[EMAIL PROTECTED]' does not exist Also, this message keeps popping up even when calls aren't going through: Sep 7 12:15:12 NOTICE[14330]: pbx.c:1688 pbx_extension_helper: Cannot find extension context 'default' On 9/7/05, Flobi [EMAIL PROTECTED] wrote: It states that the conf file overrides the static db info, but what about the ael file? Does that override also? BTW, RealTime Static...talk about oxymoron :-) Gotta love it! Flobi On 9/7/05, Matthew Boehm [EMAIL PROTECTED] wrote: CVS HEAD/Asterisk 1.2: Is there a way to have the entire extensions.conffile coming from the realtime? Yes. Go read the wiki on RealTime Static. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Automated Signature: This message is from Flobi of Flobi.com. Visit my website if you like: http://www.flobi.com/ Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- -- Automated Signature: This message is from Flobi of Flobi.com. Visit my website if you like: http://www.flobi.com/ Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- -- Automated Signature: This message is from Flobi of Flobi.com. Visit my website if you like: http://www.flobi.com/ Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- -- Automated Signature: This message is from Flobi of Flobi.com. Visit my website if you like: http://www.flobi.com/ Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth
Re: [Asterisk-Users] PHP and ASterisk Manager
Anton Krall wrote:
Guys, is anybody using PHP sockets to connect to the Manager and send
command like show voicemail users for example or any other?
My question is, how to parse the return info in a way that can be shown back
to the user via web (discard all the manager responses not needed)?
Use preg_match() to match the lines you want the user to see on the website.
$socket = fsockopen(localhost,5038, $errno, $errstr, 30);
if(!$socket) {
print No socket;
exit();
}
fputs($socket, Action: Login\r\n);
fputs($socket, Events: Off\r\n);
fputs($socket, UserName: bleh\r\n);
fputs($socket, Secret: bleh\r\n\r\n);
fputs($socket, Action: Command\r\n);
fputs($socket, Command: show channels\r\n\r\n);
fputs($socket, Action: Logoff\r\n\r\n);
while(!feof($socket)) {
$buff = fgets($socket,1024);
if(preg_match(/SIP\/.*/, $buff)) {
print I found a SIP call;
}
}
___
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Re: [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration
MySQL RealTime Static seems to see the settings as it goes through and does the select.. but the it just kinda ignores them Strange. Have you verified this behavior with ODBC RealTime? The code that parses the results is virtually identical so I don't see this as a mysql-rt specific issue. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 upgrades
You cannot go from 5.3 - 7.5. You must go from 5.3 - 7.0 then to 7.5. -Matthew From: Sascha Ferley [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 5 Sep 2005 13:19:40 -0600 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 upgrades Hi, I got a problem of having to upgrade 35 Cisco 7960 phones from default firmware of 3.1 to 7.5. The problem I get is that when trying to upgrade I see on the tftplog that it can't seem to find the file (8 character issue). So I renamed the files to suit what is supposed to be in them. I am trying incremental upgrades from 3.1 - 5.3 - 7.5, with no luck. It goes to Upgrading Software and sits there endlessly redownloading the same file. It seems to stall going no-where .. Any one successfully upgraded the phones from default? What are any of the specifics. With the 5x series do I need the P003-05 in the OS79XX.TXT file or still the P0S3 ? Anyone have any ideas as to what I should do? I can't seem to get the pre 5x versions of the software any more. Seems with my contract I can only downgrade to 5x series. All that shows on the Cisco CCO site. Please let me know Thanks Sascha ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Real-Time Voicemail Configuration
You must store voicemail.conf using RealTime Static in order to use the options you have mentioned from database. -Matthew From: Matt [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sun, 4 Sep 2005 19:51:36 -0400 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Real-Time Voicemail Configuration Hi, When using asterisk real-time with mysql voicemail integration... where exactly do I put the options like the [PBX] tag, and how long silence can be, etc? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration
How did you convert your voicemail.conf file into RT Static? Did you use the perl script? -Matthew From: Matt [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sun, 4 Sep 2005 20:37:34 -0400 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Asterisk Real-Time Voicemail Configuration I should add to this... I understand to make the table.. but when I make it.. asterisk selects it but seems to ignore things. No where have I found documented what the var_category and such are... what numbers do I put in there?!?! On 9/4/05, Matt [EMAIL PROTECTED] wrote: Hi, When using asterisk real-time with mysql voicemail integration... where exactly do I put the options like the [PBX] tag, and how long silence can be, etc? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for better Follow Me
Hauke Zuehl wrote: Hi everybody :) I am a new member here and hope that someone gives me a hint for my problem: Let's say I am at work and my SIP phone (KPhone in my case) is connected to my private Asterisk. I want to call my wife at home so her SIP phone rings. She does not pick up the phone (maybe she is somewhere in the house and has to run to the phone) so after 15 seconds her cell phone should ring. Until now it is a classic follow me but what I want: I want both phones (SIP and cell) ringing and if one phone is picked up the other phone should stop ringing. This is easially accomplished via the following single line: exten = 3044,1,Dial(SIP/herphoneZAP/R1d/5551212,60) Go read about app_dial on the wiki; it has lots of features built into it. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI Identity Crisis
Just upgraded to most recent CVS and now I get this: pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. when starting asterisk. Needless to say, I can't run asterisk without my PRI. Guess I'll start reverting code backwards day by day until I find the code that works. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI Identity Crisis
Damon Estep wrote: Sounds like you are connected to a pbx instead of a carrier switch, either asterisk or the pbx need to be set to network Pri_cpe or pri_net in asterisk... Oh, you are good. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] odbc realtime update problem
Julian Lyndon-Smith wrote: After an afternoon of chasing all sorts of dead-ends (permissions etc) I finally changed the uniqueid from an int to a character field, and it all updates ok now. Now, is this a problem with res_odbc, the linux odbc client or the sql server itself ? Must be cause I am using res_config_mysql and my uniqueid field is an INT and everything works great. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.2beta and PRI and CDR Corruption
Anyone out there running 1.2beta with a PRI and having CDR problems? I just upgraded to most recent everything and now my CDR's look like this: ,,9035646130,copper_routing,,Zap/65-1,SIP/netl-a3ac,Dial,SIP/[EMAIL PROTECTED]|60,2005-08-31 13:03:09,2005-08-31 13:03:20, 2005-08-31 13:03:29,20,9,ANSWERED,DOCUMENTATION That is a direct text copy/paste from Master.csv This only seems to affect incomming PRI calls. All other calls (inc SIP, out SIP, out PRI) show correct CDRs. I'm worried this corruption of data may eventualty lead to a crash but so far nothing. CallerIDName is parsed correctly: ,p^,8322008630,macro-faxrecordvoicemail,BOEHM MATTHEW p^,Zap/2-1,SIP/3044-b976,Dial,SIP/3044|10|wt,2005-08-31 13:06:36,2005-08-31 13:06:36,2005-08-31 13:06:42,6,6,ANSWERED,DOCUMENTATION but CallerIDNumber is not; as evidenced above. Any thoughts? Ideas? Should I be reverting asterisk code backwards or libpri? -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing Context being mistake for dialplan?
Here is one that happens at random. Probably 1 out of 1000 calls will show this: -- AGI Script Executing Application: (Dial) Options: (SIP/[EMAIL PROTECTED]|60) -- Called [EMAIL PROTECTED] Aug 31 18:27:28 NOTICE[25965]: pbx.c:1681 pbx_extension_helper: Cannot find extension context 'xocommunications' -- SIP/xocommunications-a5f5 is ringing The notice is correct, I have no extension context called xocommunications. I do have a SIP Peer called that though. Somehow it's getting confused. Anyone else seen this before? Like I said, I could have 1000 or more calls go out just like this one with no problems. Just get this one random -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime and include
Luca Lafranchi wrote: I'm interested for this thread, can you explain with an example please? In my extensions.conf I have ... [sip.proxy.com] switch = Realtime/[EMAIL PROTECTED] in extensions table on mysql I can insert on app field the command include and in the appdata field my context ? Luca Go read: http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Static -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime Queues and Agents
Julian Lyndon-Smith wrote: We use agents and queues, with CVS HEAD. I've read up on realtime queues and queue members, (and actually understand it!) but there is no reference to agents. Is it possible to have realtime agents as well ? Julian. No there isn't. And there won't be until RealTime gets updated with 'INSERT' and DELETE abilities. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime Queues and Agents
Julian Lyndon-Smith wrote: That's a bugger. Forgive me for asking, but how is is possible to be able to have SIP realtime (adding new sip phones in without having to reload) but we can't have agent realtime ? In my simple mind I substitute agent for SIP and can't compute :) because you don't add new sip phones via RealTime. you add them outside the realm of asterisk into your database. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime and include
Urban wrote: Hi, is there any support for include statement in the database when using realtime configurations? I would like to have as much as possible configuration in my postgres db but we have different access controls for different user contexts (allow international, national etc). Today we have different contexts for access rules e.g. [allow_international] exten = _00.,1,Dial... and for users we just include the allow_xxx and deny_xxx contexts. This makes it easier since we don't need to change each users dialplan just include the right contexts. Is this possible with realtime? The only way I see is to add/remove switch statements in extensions.conf and then we back to make the changes in extensions.conf and not in the database... If you store the extensions.conf in database, then it will work. If you want to use the switch, then no. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] storing voice messages in DB SQL
Its the same syntax for every other config. Just look at every other config option and replicate. Odbctable=mytablename Or Odbctable = mytablename -Matthew From: harry gaillac [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sun, 28 Aug 2005 12:11:07 +0200 (CEST) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] storing voice messages in DB SQL hello, According to docs/README.odbcstorage how can we set : /// The database name (from /etc/asterisk/res_odbc.conf) is in the odbcstorage variable in the general section of voicemail.conf. You may modify the voicemessages table name by using odbctable=??? in voicemail.conf /// what's the right syntax in voicemail.conf ? Harry --- Matthew Boehm [EMAIL PROTECTED] a écrit : Yes. Look in the apps/Makefile for USE_ODBC_STORAGE and read in the docs/ for a table structure. Right now it is ODBC only. -Matthew From: harry gaillac [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sat, 27 Aug 2005 16:12:09 +0200 (CEST) To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] storing voice messages in DB SQL Hello, Can we store voice messages in a database instead of files. Regards ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] storing voice messages in DB SQL
Yes. Look in the apps/Makefile for USE_ODBC_STORAGE and read in the docs/ for a table structure. Right now it is ODBC only. -Matthew From: harry gaillac [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sat, 27 Aug 2005 16:12:09 +0200 (CEST) To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] storing voice messages in DB SQL Hello, Can we store voice messages in a database instead of files. Regards ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime sip channel configuration - insecure option
Billy wrote: `insecure` varchar(4) default NULL, This can be changed. I just read the chan_sip.c code and the following values are acceptable: very yes true basically anything with true/false value port invite port,invite invite,port The varchar(4) was originally intended for: very, yes or no -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime and database structure
Pablo Ezequiel Fernández wrote: Is there any official structure for the tables for doing realtime sip and extensions ? I see there's a big (big in lot's of fields) on http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Sip and I've seen other partial tables in other places. Thanks. The only fields that are required are: name ipaddr port regseconds The rest are all option as per sip.conf. If it is optional in sip.conf, then it is option in realtime. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RealWorld Stats; Not achieving expected results
Hey guys, We have a brand new Dell Poweredge 1850, Single Proc 3Ghz, 2GB RAM, 15K RPM HD's RAID 1. We also have a Sangoma 4 port T1/PRI card. We are not using G729. Everything is G711. Every call is PRI - Asterisk G711 - Sip Carrier We just filled up 2 PRI's and reached a CPU usage of 70%. There's no way we can do 4 PRI's worth of traffic. Will a 2nd CPU really make that much difference? Why is there so much CPU used going from PRI to 711? I could understand 70% usage if I was using 729 but I'm not. I was expecting to be able to process 96 calls (all 4 PRIs) with about 50% (or less) CPU usage. Any clues? Suggestions? Thanks, Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Are you using a Lucent?
Is anyone out there using Lucent brand equipment to handle an incomming DS3, converting all 672 calls to SIP (as G729) and sending those to Asterisk/SER over ethernet? If you are and are willing to speak to my boss about your experiences (over the phone) with it, please contact me off list. We have a possible contract with a local CLEC to handle their long distance, and they want to send to us using DS3 and SS7. I'm trying to convince my boss to use a $9K Lucent, but he wants to spend much more by breaking out the DS3 into DS1's and stack up 6 asterisk boxes with 1 4-port card in each. Again, if you are using Lucent and are willing to speak to my boss about your experiences, please contact me off list so I can setup a call. Thanks, Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RealTime ignoringswitch=Realtime/[EMAIL PROTECTED] me_ext
Araba, Michael wrote: mail*CLI realtime mysql status Aug 24 00:56:50 ERROR[963]: res_config_mysql.c:596 mysql_reconnect: MySQL RealTime: Failed to connect database server RealTimeMaster on localhost. Check debug for more info. Aug 24 00:56:50 DEBUG[963]: res_config_mysql.c:597 mysql_reconnect: MySQL RealTime: Cannot Connect: Can't connect to local MySQL server through socket '/tmp/mysql.sock' (2) Post your res_mysql.conf. Obviously you cannot login to your mysql box for some reason. Can you connect to mysql from your shell? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RealTime ignoringswitch=Realtime/[EMAIL PROTECTED]
cdr_addon_mysql.c:264 my_load_module: Unable to load config for mysql CDR's: cdr_mysql.conf This is a problem. Where are your config files? I don't have res_mysql.conf in /etc/asterisk/. Well, there is your problem again. Why don't you have config files? You can't expect res_config_mysql to use the res_config_odbc config file can you? There are sample configs for both of these modules. How come you didn't install them? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RealTime ignoringswitch=Realtime/[EMAIL PROTECTED] me_ext
Araba, Michael wrote: I am using the current HEAD of asterisk and for asterisk-addons. I have been trying to setup realtime mysql voicemail but no sucess. I keep getting this error below. The necessary modules are loaded, res_config_mysql.so ... There is no point to repeating what you just posted. res_mysql.conf settings [general] dbhost = localhost dbname = RealTimeMaster dbuser = xxx dbpass = xxx dbport = 3306 dbsock = /tmp/mysql.sock Does /tmp/mysql.sock exist? I don't think it does since the error you get tells you this. Where is your mysql socket? Is mysql even running on this machine? extcconf.conf This file looks correct. Make sure the name of the file is correct: extconfig.conf Aug 24 00:56:50 DEBUG[963]: res_config_mysql.c:597 mysql_reconnect: MySQL RealTime: Cannot Connect: Can't connect to local MySQL server through socket '/tmp/mysql.sock' (2) This error is quite self-explanatory. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RealTime ignoringswitch=Realtime/[EMAIL PROTECTED] me_ext
John Novack wrote: I also experienced some difficulty in connecting to mysql, though for CDR, using HEAD from 2/24/05 and finally discovered that the CDR_mysql.conf wanted the host NAME in hostname, while res_mysql.conf wanted the IP address. Both Asterisk and mysql on the same machine, running RH9 Both modules should be able to use either hostname, IP, or local socket. They both use the same mysql API code to connect to the server. If mysql and asterisk are on the same machine, why aren't you using sockets? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RealTime ignoringswitch=Realtime/[EMAIL PROTECTED] me_ext
John Novack wrote: Both modules should be able to use either hostname, IP, or local socket. They both use the same mysql API code to connect to the server. Should perhaps, but they DON'T! I will test this but again, they use the same mysql_real_connect() to connect to the server. Why isn't there a good example of that on the Wiki or in the docs? If you understand how to do that, then TEACH others! It says it quite plainly in all the sample configs that if you are running a database on the same machine that you can use sockets. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lots of console; attach and grep?
We have recently started routing about 3 PRI's worth of traffic thru our asterisk box. The text on the console now flys by so damn fast, I can't really see what the heck is going on. Even with verbosity 0 and debug 0 it is still so fast. Is there some way I can attach to the console in a way that will allow me to grep or otherwise filter the text so I can focus on something in particular? Thanks, -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lots of console; attach and grep?
Eric Wieling aka ManxPower wrote: You mean like the info in /var/log/asterisk which is configured via /etc/asterisk/logger.conf ? Damn. If I change any logging, that's going to require an asterisk restart isn't it? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sql Realtime
Jimmy Smith wrote: .. options i found would be to use mysql from the asterisk distro.. but are the memory leaks fixed ? Options are to use res_config_mysql found in asterisk-addons. What memory leaks? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold + canreinvite=yes
Ronald Voermans wrote: For canreinvite=yes to work, I think I need to remove the t argument in the Dial(SIP/ext|60|t) application. Otherwise, Asterisk will allways stay in the middle. I don't want that, so I removed the 't' argument. That works. Now, when two UA are calling, Asterisk gets out of the RTP stream. However, when removing the 't' argument, the Music On Hold doesn't work anymore between these two UA. If I put one UA on hold, Asterisk states that it is starting Music On Hold, but the holding party doesn't hear the audio stream. Umm.. DUH! If you remove the RTP stream from asterisk, asterisk can't send audio (the rtp stream) to the phones. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk+realtime
Kamran Ahmad wrote: hello i m using asterisk-1.0.9. Come on people. Pay attention. What does the very first opening paragraph say: http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold + canreinvite=yes
Kevin P. Fleming wrote: Matthew Boehm wrote: Umm.. DUH! If you remove the RTP stream from asterisk, asterisk can't send audio (the rtp stream) to the phones. Umm. DUH! Yes it can. When a SIP endpoint is placed on hold, Asterisk will re-INVITE the audio stream back to itself for precisely that reason. Hmm..I stand corrected. And now that I think about it, it seems I jumped the gun without thinking. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime sip_buddies register= how?
You can store your entire sip.conf using RealTime. That should allow for register = to work. -Matthew From: Guillermo Krepper [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sat, 20 Aug 2005 13:05:02 +0200 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Realtime sip_buddies register= how? Hi all I've been doing some testing on realtime using mysql, an have a little question that could not find the answer to or maybe its not posible at this time. Is there a way use register=.. on a DB using realtime. For the moment I use it in sip.conf. It will help me a lot if this could be store on a DB somehow. commets or sugestions ? thanks Billy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why do I get pbx.c 1645 pbx_extension_helper: No application 'Voicemailman' for extension
You spelled Voicemailmain wrong somewhere. Or your extensions are not in sync with the conf file. Verify that the extensions.conf is correct then 'extensions reload'. You can also do show dialplan context to view what is currently loaded in memory. -Matthew From: Angus Comber [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sat, 20 Aug 2005 19:21:07 +0100 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Why do I get pbx.c 1645 pbx_extension_helper: No application 'Voicemailman' for extension Does VoicemailMan have to be installed ? Why not available. I have setup a mailbox in voicemail.conf and I can leave a voicemail - just cannot pickup up using *97. My *97 code in extensions.conf: exten = *97,1,Answer exten = *97,2,VoicemailMain([EMAIL PROTECTED]) exten = *97,3,Hangup asterisk console: Verbosity was 8 and is now 12 -- Executing Answer(SIP/200-d83a, ) in new stack Aug 20 18:57:45 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No application 'VoicemailMan' for extension (default, *97, 2) == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-d83a' -- Executing Answer(SIP/200-81f6, ) in new stack Aug 20 18:57:59 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No application 'VoicemailMan' for extension (default, *97, 2) == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-81f6' -- Executing Answer(SIP/201-a86c, ) in new stack Aug 20 19:00:24 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No application 'VoicemailMan' for extension (default, *97, 2) == Spawn extension (default, *97, 2) exited non-zero on 'SIP/201-a86c' -- Executing Dial(SIP/201-1e08, SIP/200|20|Ttm) in new stack -- Called 200 -- Started music on hold, class 'default', on SIP/201-1e08 -- SIP/200-b925 is ringing -- Stopped music on hold on SIP/201-1e08 -- Nobody picked up in 2 ms -- Executing VoiceMail(SIP/201-1e08, su200) in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/200/INBOX/msg format: wav49, 0x818eb40 -- x=1, open writing: /var/spool/asterisk/voicemail/default/200/INBOX/msg format: gsm, 0x813a7e8 -- x=2, open writing: /var/spool/asterisk/voicemail/default/200/INBOX/msg format: wav, 0x818ed88 -- User hung up == Spawn extension (default, 200, 2) exited non-zero on 'SIP/201-1e08' -- Executing Answer(SIP/200-4b1a, ) in new stack Aug 20 19:01:57 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No application 'VoicemailMan' for extension (default, *97, 2) == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-4b1a' -- Executing Answer(SIP/200-5369, ) in new stack Aug 20 19:02:11 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No application 'VoicemailMan' for extension (default, *97, 2) == Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-5369' linux*CLI Angus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Out of G.729 Decoder Licenses!
Innocent Evil wrote: Hello, I have SIP and Asterisk. On Asterisk I have 2 liscenses of g729 (from digium website) SIP user (100) is calling another SIP user (101). As 101 is not online, my SIP server is redirecting that call to Asterisk. Asterisk forward it to 101's voice mail box. SIP user 100's phone have g729 codec. I havn't buy any codec for SIP server itself. But when 100 reach at 101's voice mail, I get this: Aug 18 18:31:36 WARNING[14511]: codec_g729.c:180 g729tolin_framein: Out of G.729 Decoder Licenses! I didn't get it. Would anybody please explain it. Are the licenses installed? Do show g729 from CLI. You will need a g729 license to access asterisk voicemail from a g729 phone. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNMP for Asterisk
Stojan Sljivic - GDS wrote: Hi Harry, I have tried to install it, but it gives compilation errors on Fedora Core 3. Also, patch file for Asterisk was not in sync with version of Asterisk I had, but I managed to apply changes manually. Did anyone succeeded to run that package on FC3 or Red Hat Enterprise 3? I am currently working on a new updated SNMP module for Asterisk. It is currently in testing stage as I am having a problem with channel listings. It will support read-only attributes associated with Asterisk such as number channels, number of calls, all config file paths, all modules, all apps, etc.. I am hoping to have it ready for inclusion in the 1.2 release. Once my internal testers have proven it worthy, I will release to public for more testing. Watch for it. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How real time is realtime?
Asterisk Supporter wrote: How real time is realtime? If the extensions.conf is stored in the database, does * query it row by row or is it cached? In other words, given the following exerpt: If you store the .conf file using realtime, then yes it is cached. If you simply use realtime extensions, then no it is not and you can change priority 2 while the call is at 1 and have it take effect immediatly. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk configuration from database with res_config
That wiki page is old, ugly and out of date. There are many like it and if I only knew how to delete wiki pages, I would clean it up some. The easiest way, Frank, to do what you want is to download CVS-HEAD and use ARA to store your config files. Also download addons from HEAD and you can use the native mysql realtime driver. http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime -Matthew Frank Aartman wrote: I want to let Asterisk read its configuration from a mysql database. I configured everything according to the wiki page: http://www.voip-info.org/tiki-index.php?page=Asterisk+res_config. However it doesn't work. I am using 1.0.8 asterisk version and here are my config files: Extconfig.conf: [settings] ;uncomment to load queues.conf via the db engine. ;queues.conf = odbc,mysql1,ast_config ;extensions.conf = odbc,mysql1,ast_config sip.conf = odbc,mysql1,ast_config res_odbc.conf: ;;; odbc setup file [mysql1] dsn = MySQL-asterisk username = blaat password = blaat pre-connect = yes [mysql2] dsn = MySQL-asterisk username = myuser password = mypass pre-connect = yes odbc to mysql is working fine, I tested it. here is my odbc.ini from /etc/ [MySQL-asterisk] Description = MySQL Asterisk database Trace = Off TraceFile = stderr Driver = MySQL SERVER = localhost USER= blaat PASSWORD= blaat PORT= 3306 DATABASE= asterisk I used the load_res_config.pl to put the sip.conf into the database in ast_config. Via phpMyadmin I can see the data in there correctly. When booting or reloading Asterisk I don't see anything indicating it is connecting to odbc. I tried removing the sip.conf from /etc/asterisk, leaving an empty sip.conf, and only leaving the general section of sip.conf there. Nothing works. Cheers, Frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk with odbc
Have you looked in the debug log for any errors? Have you checked the ODBC connection using Asterisk CLI? I believe if you type 'odbc ?' then you will see a command to check the status of a connection. -Matthew Kamran Ahmad wrote: hello i am trying to use res_odbc for sipuser. my connection is working. i have checked using isql. even cdr_odbc is working but i hav problem in res_odbc. i have created user in sip_buddies table but asterisk is no getting user from this sip_buddies table. /etc/asterisk/extconfig.conf [settings] sipusers=odbc,asterisk,sip_buddies sippeers=odbc,asterisk,sip_buddies /etc/asterisk/res_odbc.conf [asteirsk] dsn=asteriskdsn username=voipbilling password=voipbilling pre-connect=yes ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNMP for Asterisk
Stojan Sljivic - GDS wrote: Hi Matthew, Nice to hear that. What will be the license type for your SNMP module? Are you going to include it in the Asterisk CVS or it will be independent product? Regards, Stojan Sljivic Same as asterisk. It will most likely be in asterisk-addons. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNMP for Asterisk
harry gaillac wrote: Matthew Could you tell us the differences between your project and ast-ax-snmpd . Regards Harry Not really many differences at all. It's basically Andrea's code cleaned up. His code was scattered across about 10 different files/headers. I combined them into one module; cleaned up the MIB to be compliant with current spec; tweaked some code here and there. Right now, I did remove all the zap specific stuff. I will add it back in if people want it. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNMP for Asterisk
Stojan Sljivic - GDS wrote: Hi Matthew, Are you using ucd-snmp or net-snmp? Regards, Stojan Sljivic UCD is deprecated, its should be net-snmp. I don't include any of the ucd headers. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] options for mysql query from dialplan
Hi Damon, You are basically doing EXACTLY what we are doing right now; except we are doing more. We now have an AGI PHP script that does the following for every call: - Connect to MySQL over LAN - If the dialed number begins with 1, strip it. - SELECT State FROM lcr_lata WHERE NPA = $dial_npa AND NXX = $dial_nxx - Do some PHP logic to determine if Interstate vs Intrastate - SELECT rate, address, technology, prefixes FROM lcr_rates LEFT JOIN lcr_carriers USING(carrierid) WHERE NPA = $dial_npa AND NXX = $dial_nxx AND carrier_active = 1 ORDER BY rate ASC; - Loop thru results. lcr_rates has 329,530 rows. lcr_carriers has 8 rows. lcr_lata has over 150,000 rows. Everything preforms in real time. Here is a sample query of a call that just went thru: SELECT r.Interstate, rc.name, rc.technology, rc.address, rc.prefix FROM lcr_rates r LEFT JOIN lcr_carriers rc ON r.CarrierId = rc.id WHERE r.NPA = '254' AND r.NXX = '463' AND r.active = 1 ORDER BY r.Intrastate ASC, r.NPA DESC, r.NXX DESC Query took 0.0025 sec. I don't see how your table with 300K rows is preforming worse than ours. You got indexes? To make this even better, our MySQL server is a Quad P3 500 Mhz machine. Works great here. -Matthew Damon Estep wrote: I am using realtime mysql for extensions, sip, and voicemail. Outbound call routing does not really perform well in realtime extensions due to the high number of rows in the database (300k), so I can not use it. It appears with my limited knowledge that the query method is not robust enough for large databases. Given the fact that I already have realtime and mysql configured, what are my options for running a mysql query from the dialplan to find the provider I want to use for outbound. I am not looking for a complete solution, just a hint on the best way to query my existing mysql database from the dialplan. I have looked at the MySQL command, and there are a lot of notes about connection closing and other scary stuff? Does it work? Are there other native options given the fact that realtime is configured and in use? The goal is to run a query against a database like this SELECT provideralias FROM ldproviders WHERE npa = (digits 2 thru 4 of dialed number) AND nxx = (digits 5 thru 7) Then take the provider alias returned and Dial(SIP/[EMAIL PROTECTED],60). Next step would be to add a loop for multiple providers, starting with the lowest cost. Any hints or comments from the pros? TIA Damon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which AGI Development Software is fastest on Asterisk?
Asterisk wrote: I'm looking to develop some custom AGI that will be MySQL intensive. It appears Asterisk supports many different development environments. Which would be best suited for Asterisk and MySQL? Bart I use PHP. Love it. Fast, Easy. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime caching
We have a web interface where users can update their dialplan online (not in production yet). The web page modifies the mySQL record. It seems that some options are not re-read when caching is on, for example, changing the caller ID value in the sip table has no effect until a reload (or expiration), so at least in some cases rtcahcefriends makes realtime notsorealtime. No. It is doing exactly what it says it will, cacheing. If you have rtcachefriends turned on, when a peer/user registers the info is pulled from DB and added to the internal (a la 'in memory') list that chan_sip maintains. If you change something in DB after this occurs then your changes won't take affect because chan_sip has no need to re-lookup your phones info since the info is already present in memory. What you can do is use sip prune realtime name to remove just the single peer/user from memory. And you can force a reload of that peer from realtime by using sip show peer name load. If you want pure realtime where chan_sip always pulls from db, then turn caching off. Keep in mind that turning caching off will remove MWI and NAT functionality. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime caching
Damon Estep wrote: I may have answered my own question, is it true that realtime extensions are still queried every call, and only chan_sip is effected by rtcachefriends? Damon True. RealTime Exensions are queried every time. There is no caching of extensions. If you turn on debug log, you can watch each query. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime caching
Jimmy Smith wrote: pruning breaks asterisk on high loads at least on all 5 of our servers. all using different versions and custom. You should bug report this if you have a backtrace. Kevin and I worked on the pruning stuff (well, he coded and i tested) for a while and seemedly got it working. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail file permissions
Tzafrir Cohen wrote: On Wed, Aug 17, 2005 at 07:48:29AM -0400, hugolivude wrote: Is there a way around this w/o giving everyone root privileges! Run asterisk as its own user/group. We do. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quad t1 / 1U rack server combos
Damon Estep wrote: What 1u server combos work with the new quad pri cards UNDER LOAD (more than 75% channel use). Every user that buys a Digium PRI card should not have to play hit or miss with 2 or 3 servers that cost more than the card to get it to work. We use a Sangoma 4 port T1 card in our Dell Poweredge 1850 (1U) and it works like a champ. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime caching
I have reviewed the info below from the sip.sample.conf, but I must be dense, still don’t get it. flips on tv to the asterisk televangelist channel Do you find the RealTime comments in sip.conf just a little too confusing? Are you frustrated by the use of double negatives in configuration options? Do not be afraid. You are not alone. Follow the path to enlightenment and visit: http://bugs.digium.com/view.php?id=4075; It is my understating that removing rtcachefriends will break MWI? Is that true? Yes. What exactly are you trying to accomplish? Are your peers/users not being updated in your database? Are you sure? Are you watching debug for SQL log? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Asterisk Installations + SER
Brian Capouch wrote: I'm trying to figure out how to do some things like round-robin server balancing and the like using Realtime, and it seems like the right way to do it would be either via pre-processing the SQL requests coming in, or using stored procedures in the database that would accomplish the same thing. Not quite sure I understand the need to pre-process SQL requests. We just run 1 MySQL server on a RAID5 array. 1 webserver running the scripts which allow the customers to login and do their configuring. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Asterisk Installations + SER
Ronald Voermans wrote: I'm not sure I understand what you mean... I want to have internal extensions (100, 101, 102, etc.) and some full phone-numbers (10 digits). How do I implement this in *? Ronald Right. We have 3 contexts. 1 is for all incomming traffic from PRI or other carrier. 1 for each company and 1 for all outbound. For us, each company has their own context. This context handles all extensions local to the company. (eg, 100, 101, etc). Then you have a pattern match for when the company dials a 10+ digit number. (or 9 followed by any number of digits) We send all these outbound numbers from all company contexts to a master context for outboud dialing (using Goto). This 'master outbound' context actually includes the 'master incomming' context as part of its dialplan so if any customer dials the 10 digit number of another company, the call stays within asterisk, completly SIP. If no match is found, the call is sent thru a PHP script for least cost routing out via local PRI or SIP carrier. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo calibration with ztmonitor and a test line from a telco
I have been doing a bit of this too lately. This was also useful. http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html Dan What about for PRI lines? We get echo every now and then. The docs link above references FXO lines. We have none. But we do have 4 PRIs. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Asterisk Installations + SER
- I've installed multiple asterisk instances on one server (via vserver). Each * is for one customer, and has it's own extensions (like 100, 101, 102, etc.) Note that the same extension can exist on other * instances This is completely UNNECESSARY if you simply use contexts. We have 1 asterisk server running 6 different companies and a good majority of their extensions overlap. This is very easy to configure. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Asterisk Installations + SER
Do you also have some kind of tool so the companies can manage their own context ? Yup. We use RealTime Extensions. Customers login to a password protected website. User/pass is tied to their context so they are able to add/delete/update anything in their extensions context. We tell them that extensions managing is a highly advanced tool. If they screw it up, they get charged a fee for us to fix it. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_voicemail.c still looking for config fileeven I try to configure the voicemail from database.
Wei Kun wrote: mysql select * from extensions_table; ++--+---+--+---++ | id | context | exten | priority | app | appdata| ++--+---+--+---++ | 1 | from-sip | 2000 |1 | Dial | SIP/2000|20| | 2 | from-sip | 2000 |2 | Voicemail | u2000 | You need to call voicemail app with a context. Change the line above to show [EMAIL PROTECTED] -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] list in asterisk cli is getting too long
Hilton Williams wrote: - Original Message - From: Ronald Wiplinger To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, August 12, 2005 6:04 AM Subject: [Asterisk-Users] list in asterisk cli is getting too long How can I use something like|morein CLI ? The lists are getting too long, like sip show users You can also use sip show peers like pattern to truncate the list. Ex: sip show peers like 300 will only show peers whos username starts with or contains 300. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_voicemail.c still looking for config file even I try to configure the voicemail from database.
Wei Kun wrote: Hi I am trying to make asterisk load config from database, so far I get the sip, extension working, but voicemail seems still looking for config file, not from the database. Aug 11 15:04:08 WARNING[9316] app_voicemail.c: No entry in voicemail config file for '2001' How exactly are you calling it? Are you specifying the right voicemail context? What did the debug log say? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime + MYSQL
Timur V. Elzhov wrote: So the correct line in extconfig.conf must be voicemail = mysql,asterisk,voicemail_users Yes, Timur is correct. By stating that you want to bind voicemail.conf you mean you want to store the config file itself. This is not what you are looking for. Change the line above to what Timur says and it should work fine. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 79XX and VLANS
Hey gang, We have about 30 Cisco 79XX phones all running latest 7.5 SIP. We are also using all Cisco Switches and Routers. Everything works great except that when you reboot a phone it takes like 3-5 minutes for it to come up. The phones spend tons of time 'Configuring VLAN..' We don't run any VLANs. Is there some way to skip this? In the 'Network Settings' I have both 'Operational VLAN Id' and 'Admin VLAN Id' set to blank values. Any ideas? Thanks, Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
Justin Selleck wrote: Is asterisk 2.0 real? Running in c#? I see references to it but cannot find it anywhere. No. It's not out. 1.2 isn't even out. And thank god its NOT programmed in C#. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime odbc/mysql eating connections
Since you are using ODBC, this seems more likely to be an ODBC issue. If you are concerned, you should just use the native MySQL RealTime driver. It does not exibit the behavior you mentioned. -Matthew Frank Sautter wrote: our asterisk is configured to retrieve sippeers and iaxpeers via odbc from a mysql database. after each call show processlist; within the mysql console shows 2 more persistent connections which are showing no further activity and will not go away even after restaring asterisk. is anybody else experiencing this? what can i do do resolve this? this is a show processlist on the mysql console +-+--+---+--+-+---+---++ | Id | User | Host | db | Command | Time | State | Info +-+--+---+--+-+---+---++ | 7 | asterisk | localhost | asterisk | Sleep | 2 | | NULL | 8 | asterisk | localhost | asterisk | Sleep | 13596 | | NULL| 11 | asterisk | localhost | asterisk | Sleep | 13596 | | NULL . stuff deleted ... | 171 | asterisk | localhost | asterisk | Sleep | 31| | NULL | 172 | asterisk | localhost | asterisk | Sleep | 31| | NULL | 173 | asterisk | localhost | asterisk | Sleep | 1 | | NULL | 174 | asterisk | localhost | asterisk | Sleep | 1 | | NULL +-+--+---+--+-+---+---++ 160 rows in set (0.00 sec) # less /etc/odbc.ini [asterisk] Description = MySQL Asterisk database Trace = Off TraceFile = stderr Driver = MySQL Socket = /var/run/mysqld/mysqld.sock Server = localhost User= asterisk Password= obscured #Port = 3306 Database= asterisk # less /etc/asterisk/res_odbc.conf [asterisk] dsn = asterisk username = asterisk password = obscured pre-connect = yes # less /etc/asterisk/extconfig.conf [settings] iaxusers = odbc,asterisk,iaxfriends iaxpeers = odbc,asterisk,iaxfriends sipusers = odbc,asterisk,sipfriends sippeers = odbc,asterisk,sipfriends ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mysql sock location
Wei Kun wrote: Hi; In case your * and mysql are running on the same machine, and you get error Failed to connect to mysql database server ... when using Asterisk with Mysql database, check the location of mysql.sock not /tmp/mysql.sock, but /var/lib/mysql/mysql.sock Regards Kun The location of your mysql.sock is completly configurable when you first install mysql. ./configure --with-sock=/some/path/to/mysql.sock If you run with the defaults, yea it won't goto /tmp/mysql.sock Most admins don't use default. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to config voicemail with mysql?
res_mysql.conf controls the RealTime interface driver to MySQL. cdr_mysql.conf controls the MySQL CDR Addon. Are you running CVS-HEAD? Have you installed res_config_mysql.so? What happens when you type realtime mysql status ? Did you look in the debug log for errors? -Matthew Wei Kun wrote: Hi; I followed the online http://www.onlamp.com/lpt/a/3956 to configure voicemail. Now it works well with voicemail.conf and store voicemail as file. Now I want to try to test out storing voicemail within mysql database, but nothing is inserted into the table. It seems Asterisk still is following the viocemail.conf. Do I miss some config? I follow http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Voicemail In exconfig.conf voicemail = mysql,asterisk,voicemail_users And the table is in the database. The CDR can insert fine into this database, but nothing regarding voicemail. btw, what the relationship between res_mysql.conf and cdr_mysql.conf, looks like a lot of duplicated info. Thanks Kun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and PostgreSQL
Bastian Schern wrote: Hello everybody, now I'm using MySQL for SIP/IAX friends and CDR. Is it also possible to use PostreSQL instead of MySQL? Regards Bastian If you want to go thru the hassle of installing ODBC and all related stuff to run PSQL, sure you can. -Matthew P.S. stick with mysql. :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with realtime SIP
vinod malani wrote: Hi Guys, We have just set up Asterisk 1.0.7 with (CVS Head) for a realtime enviorment using MySQL Asterisk Addons. 1.0.7 is NOT CVS HEAD! 1.0.7 is STABLE and RealTime doesn't work on STABLE! -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MWI and Realtime
Michael Baird wrote: I'm testing my asterisk system and the realtime backend. My Asterisk build is rather aged, 03/18/2005 CVS. I have successfully moved Sip peers and Voicemail boxes to the realtime database backend and this works very well except for MWI. I don't seem to be able to get MWI to work when I store the voicemail information in a database backend, from a flat file it does work fine. I'm using rtcachefriends=yes for my sip users, per the WIKI, I'm presuming asterisk can't see these mailboxes, and therefore can't poll them to send the alerts when necessary. Is there anything that can be done to make this work properly, short of going back to a flat file for voicemail.conf? Regards Michael Baird We have been using RealTime and Voicemail for quite some time and have no problems getting MWI's. If your source is that old, it might be why the config option isn't working. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MWI and Realtime
turned on there. I do get the Sip Notify when a message is left, just no stutter tone when picking up the phone. I will also refresh my build as well, hopefully no big changes. The stutter tone will be a phone/ATA specific issue not related to Asterisk. For instance, I know the Linksys PAP2-NA's give you the option of stutter tone, or quick ring, etc..when there are voicemails. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Full T38 sip Faxing now Available
OK. I called the guy and this is basically the jist: Get a T38 enabled ATA. There are now many out there with firmware upgrades to enable T38. Get a Cisco 5300 (or some other gateway that supports T38). Register ATA with asterisk. Turn on reinvites for the UA. Make a call from the ata and have the call then Dial([EMAIL PROTECTED]). Asterisk will re-invite the ATA and 5300 so the audio is passed directly between them. *bing* T38 faxing. What this guy is offering is basically the use of his 5300 at 2 cents a minute to terminate. Right now he only has incomming Los Angeles abilities. -Matthew Marc Storck wrote: Do you want to share your knowledge how to get it work??? Regards, Marc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Full T38 sip Faxing now Available
Chris Mason (Lists) wrote: Asterisk will re-invite the ATA and 5300 so the audio is passed directly between them. What happens if the ata is on your private network, behind NAT. Will it still reinvite? If you have a good NAT device (like a PIX) then it should. We got 1 to go thru by doing NAT=no and canreinvite=yes on an ATA that was behind a PIX. We are trying again now to force a static public IP to an ATA and see if it works that way. Right now we can't get the re-invites to happen. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_dbodbc for asterisk stable 1.09
app_dbodbc has been publically deprecated by the author and he isn't updating it. Functionality provided by ast_data is provided by RealTime. You will need CVS-HEAD to use RealTime. Or wait a month for 1.2 to come out. -Matthew Quoting Umar Sear [EMAIL PROTECTED]: Hi, Has anyone manage to comile app_dbodbc or ast_data with the latest stable release (1.09). If so can you give some guidence on howto do it as I have trouble getting either working. Umar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ast_config not updating voicemail password
Bruce Komito wrote: I've been using realtime to store my voicemail configuration in a mysql table for several months now, and have had no problems...until today. A few weeks ago, I upgraded to the latest CVS and today I noticed voicemail is not updating the password when the user changes it through option 0. I'm not sure when this started happening, but I assume it was sometime after I upgraded. Has anyone else seen a problem like this, and if so, what's the solution? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 There was a recent patch to voicemail that removed an incorrect error. It delt with changing password with realtime. Does it say on console that voicemail password was changed? Does Allison say it was saved? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can I use MySQL in the dialplan?
What the hell? NO! show application MySql app_addon_mysql is the name of the module. load app_addon_mysql.so -Matthew Quoting Ronald Wiplinger [EMAIL PROTECTED]: Matthew Boehm wrote: Ronald_Wiplinger wrote: I would like to put / get some data from an MySQL database. I want to use this MySQL database also via a web page. bye Ronald app_addon_mysql or use RealTime. *CLI show application app_addon_mysql Your application(s) is (are) not registered I want to use it for putting stored speed dial numbers into the per phone stored register, ... I guess I cannot get that with realtime done!!! bye Ronald Wiplinger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Full T38 sip Faxing now Available
Quoting Michael D Schelin [EMAIL PROTECTED]: Hello everybody, for all of you that have searched for a real fax solution, look no further. We now have T38 faxing. Please contact me for more information. Thanks Michael D. Schelin ShellTel 626-814-2354 Biting too. Send info. -Matthew This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql does not write to mysql db
sylvain garcia wrote: Kib Eki a écrit : Asterisk don't use directly mysql database for cdr, astersisk use odbc and odbc connect to mysql. So you must configure odbc corectly wiyt libmyodbc (on debian) the config file are here: Wrong. Asterisk can and does connect to MySQL directly. (Where the hell are these people getting this wrong info?) -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cannot find channel_pvt.h
wassim darwish wrote: when i tried to compile asterisk-oh323 i get an error that channel_pvt.h is missing,where i can find and download it and in which directory i must put it. channel_pvt.h has been deprecated for quite some time. You need to update your 323 source as it shouldn't be using that header. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Real-time for H.323?
Ronald_Wiplinger wrote: Matthew, can we use real-time also for H.323 phones? (h323_buddies) ??? bye Ronald I don't see any realtime code in either H323 source. So, No. Not until someone adds it. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql does not write to mysql db
Mohamed A. Gombolaty wrote: Dear Kib, As I believe the Realtime options concerning the mysql database can only be used with the Asterisk CVS-HEAD version it's still not implemented on Asterisk v 1.0.* . Thx MAG Wrong. He's not using RealTime. No where in his original post did he mention realtime. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
