Re: [asterisk-users] Upgraded server crashes on voicemail storage
Thank you for your time. I've put my replies to your questions in-line, below. On Wednesday, June 07, 2017 10:19:41 AM Antony Stone wrote: > On Tuesday 06 June 2017 17:54:59 Mike Diehl wrote: > > > Hi all, > > > > I'm upgrading to Asterisk 13.14.0 x86_64. During my beta testing, I've > > discovered that my server crashes as soon as I leave a voicemail message. > > I'm using odbc voicemail storage as well as mysql dynamic configuration. > > > > I'm using unixODBC 2.3.2-r2 with myodbc 5.2.7-r1 > > > > I suspect that the odbc drivers are the problem. Is ther an alternative > > drive that I should be using? > > > > Failing that, any other ideas? > > Give us more details of what you mean by "crashes". My remote console gets disconnected from the Asterisk server, waits a few seconds, reconnects and shows me the start-up log. It's just like if you told asterisk to restart now. > What happens, what do you get in the Asterisk logs, what do you get in > syslog, > what state is the machine in afterwards, is there a kernel panic, what > information leads you to suspect the ODBC drivers...? What I see in the log is: [Jun 7 14:23:58] VERBOSE[11347][C-0001] app_dial.c: Everyone is busy/congested at this time (1:0/0/1) [Jun 7 14:23:58] VERBOSE[11347][C-0001] res_agi.c: magic_switch.pl: --- jmd (CHANUNAVAIL) [Jun 7 14:23:58] VERBOSE[11347][C-0001] res_agi.c: AGI Script Executing Application: (voicemail) Options: (1505903@default,su) [Jun 7 14:23:58] VERBOSE[11347][C-0001] file.c: Playing '/var/spool/asterisk/voicemail/default/15059035700/unavail.slin' (language 'en') [Jun 7 14:24:08] VERBOSE[11347][C-0001] file.c: Playing 'beep.ulaw' (language 'en') [Jun 7 14:24:09] VERBOSE[11347][C-0001] app_voicemail.c: Recording the message [Jun 7 14:24:09] VERBOSE[11347][C-0001] app.c: x=0, open writing: /var/spool/asterisk/voicemail/default/15059035700/tmp/x8hgQD format: wav, 0x7d380013d750 [Jun 7 14:24:12] VERBOSE[11347][C-0001] app.c: User ended message by pressing # [Jun 7 14:24:12] VERBOSE[11347][C-0001] file.c: Playing 'auth-thankyou.ulaw' (language 'en') [Jun 7 14:24:13] VERBOSE[11347][C-0001] config.c: Parsing '/var/spool/asterisk/voicemail/default/15059035700/INBOX/msg0004.txt': Found +++ CRASH! +++ [Jun 7 14:24:15] Asterisk 13.14.0 built by root @ voip11 on a x86_64 running Linux on 2017-06-06 21:26:05 UTC [Jun 7 14:24:15] VERBOSE[11362] config.c: Parsing '/etc/asterisk/logger.conf': Found I am thinking it's the odbc driver because I believe the server was stable before I rebuilt it with odbc voicemail storage support; it had been using the file system for storage. I'm in the process of migrating all of my servers to database storage. > Also, what have you upgraded from, what machine specs are you running on, > what's the dialplan section dealing with leaving voicemail...? The ONLY thing I changed from the previous configuration was to convert to odbc voicemail storage. > The more info you give us, the more likely it is we can suggest something > useful. Ya, I understand; I was just tired... and frustrated. Thanks again for your time. -- Mike Diehl Diehlnet Communications, LLC. Sales: (800) 254-6105 Support: (505) 903-5700 Fax: (505) 903-5701 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip: Inbound calls: selecting the correct trunk with one provider and different numbers
On Wed, Jun 7, 2017, at 05:28 PM, Michael Maier wrote: > Hello! > > I've got a problem to select the correct trunk if there is one provider > and different numbers with different configurations for this same > provider. > > Example: > > trunk-prov1-2345 > trunk-prov1-2346 > trunk-prov1-2347 > > Each trunk registers an own number (at the same provider) and provides > own configuration: they have different allowed codecs e.g.. > > What I'm experiencing now, is, that each incoming call is provided by > trunk-prov1-2346, no matter which number has been dialed. > > The problem isn't the routing (this is done on base of the correct DID), > but the problem is, that wrong codices are used if the wrong trunk is > selected. > > Is this a problem of asterisk or is it caused by the provider, which > always addresses the same "trunk" regardless which number has been > called? Asterisk is the one who associates an incoming message with an endpoint. In the case of providers you can use IP based matching - which would behave as you see, only one can be matched. The second option is the line option[1] which may or may not work (it depends on the behavior of the provider). If it works then the right endpoint would be chosen. Out of those two options there's nothing else applicable built in to match. [1] http://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/ -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pjsip: Inbound calls: selecting the correct trunk with one provider and different numbers
Hello! I've got a problem to select the correct trunk if there is one provider and different numbers with different configurations for this same provider. Example: trunk-prov1-2345 trunk-prov1-2346 trunk-prov1-2347 Each trunk registers an own number (at the same provider) and provides own configuration: they have different allowed codecs e.g.. What I'm experiencing now, is, that each incoming call is provided by trunk-prov1-2346, no matter which number has been dialed. The problem isn't the routing (this is done on base of the correct DID), but the problem is, that wrong codices are used if the wrong trunk is selected. Is this a problem of asterisk or is it caused by the provider, which always addresses the same "trunk" regardless which number has been called? Thanks for any hint, regards, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgraded server crashes on voicemail storage
On Tuesday 06 June 2017 17:54:59 Mike Diehl wrote: > Hi all, > > I'm upgrading to Asterisk 13.14.0 x86_64. During my beta testing, I've > discovered that my server crashes as soon as I leave a voicemail message. > I'm using odbc voicemail storage as well as mysql dynamic configuration. > > I'm using unixODBC 2.3.2-r2 with myodbc 5.2.7-r1 > > I suspect that the odbc drivers are the problem. Is ther an alternative > drive that I should be using? > > Failing that, any other ideas? Give us more details of what you mean by "crashes". What happens, what do you get in the Asterisk logs, what do you get in syslog, what state is the machine in afterwards, is there a kernel panic, what information leads you to suspect the ODBC drivers...? Also, what have you upgraded from, what machine specs are you running on, what's the dialplan section dealing with leaving voicemail...? The more info you give us, the more likely it is we can suggest something useful. Antony. -- This email was created using 100% recycled electrons. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
