Leandro Dardini
2016-Jul-06 15:10 UTC
[asterisk-users] Impossible to use any recent asterisk version with chan_sip
Hello, I'd like to know if anyone of you is finding my same problems using any recent asterisk version, after 13.7 / 13.8 with chan_sip. If I use any recent asterisk version, after just few seconds asterisk completely locks up, stopping processing SIP/UDP packets. Nothing is written in the asterisk log, but if I run "netstat -nap | grep 5060" I see the UDP buffer filled up. If I step back to asterisk 13.2, then all is fine and asterisk is rock solid. I know I should use PJSIP and chan_sip is no more supported, but at this point, if this is the working state of chan_sip, it should be completely removed. Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160706/592458a1/attachment-0001.html>
Joshua Colp
2016-Jul-06 15:14 UTC
[asterisk-users] Impossible to use any recent asterisk version with chan_sip
Leandro Dardini wrote:> Hello, > I'd like to know if anyone of you is finding my same problems using any > recent asterisk version, after 13.7 / 13.8 with chan_sip. > > If I use any recent asterisk version, after just few seconds asterisk > completely locks up, stopping processing SIP/UDP packets. Nothing is > written in the asterisk log, but if I run "netstat -nap | grep 5060" I > see the UDP buffer filled up. > > If I step back to asterisk 13.2, then all is fine and asterisk is rock > solid. > > I know I should use PJSIP and chan_sip is no more supported, but at this > point, if this is the working state of chan_sip, it should be completely > removed.It's not the normal working state of chan_sip, in fact it hasn't really been touched in any way that would cause this to happen and I haven't seen any bug reports along these lines. I also know that FreePBX is running both chan_sip and chan_pjsip and haven't experienced anything so it may be isolated to you. If you follow the instructions on the wiki[1] it will provide a backtrace which will show where chan_sip is hanging. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock -- 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
Barry Flanagan
2016-Jul-06 15:18 UTC
[asterisk-users] Impossible to use any recent asterisk version with chan_sip
On 6 July 2016 at 16:14, Joshua Colp <jcolp at digium.com> wrote:> Leandro Dardini wrote: > >> Hello, >> I'd like to know if anyone of you is finding my same problems using any >> recent asterisk version, after 13.7 / 13.8 with chan_sip. >> >> If I use any recent asterisk version, after just few seconds asterisk >> completely locks up, stopping processing SIP/UDP packets. Nothing is >> written in the asterisk log, but if I run "netstat -nap | grep 5060" I >> see the UDP buffer filled up. >> >> If I step back to asterisk 13.2, then all is fine and asterisk is rock >> solid. >> >> I know I should use PJSIP and chan_sip is no more supported, but at this >> point, if this is the working state of chan_sip, it should be completely >> removed. >> > > It's not the normal working state of chan_sip, in fact it hasn't really > been touched in any way that would cause this to happen and I haven't seen > any bug reports along these lines. I also know that FreePBX is running both > chan_sip and chan_pjsip and haven't experienced anything so it may be > isolated to you. If you follow the instructions on the wiki[1] it will > provide a backtrace which will show where chan_sip is hanging. > >Joshua, This was actually reported in https://issues.asterisk.org/jira/browse/ASTERISK-25468 with backtraces. It appeared to have started in 13.5, as I tested all from 13.2 to 13.6 at the time Hope this helps. -Barry Flanagan> -- > 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160706/a9c089b7/attachment.html>
Joshua Colp
2016-Jul-06 19:46 UTC
[asterisk-users] Impossible to use any recent asterisk version with chan_sip
Leandro Dardini wrote:> Hello, > I'd like to know if anyone of you is finding my same problems using any > recent asterisk version, after 13.7 / 13.8 with chan_sip. > > If I use any recent asterisk version, after just few seconds asterisk > completely locks up, stopping processing SIP/UDP packets. Nothing is > written in the asterisk log, but if I run "netstat -nap | grep 5060" I > see the UDP buffer filled up. > > If I step back to asterisk 13.2, then all is fine and asterisk is rock > solid. > > I know I should use PJSIP and chan_sip is no more supported, but at this > point, if this is the working state of chan_sip, it should be completely > removed.I've responded on the issue but the backtrace you've provided makes it appear as though the issue is actually in ODBC, which since chan_sip is using it in your deployment it causes it to lock up (why exactly is unknown). Since it's separate you should create a new issue. If you don't want to I can do so tomorrow. The complete console output (with debug going to console in logger.conf and core set debug 3) as well as the configuration would also be useful. Cheers, -- 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
Leandro Dardini
2016-Jul-06 20:48 UTC
[asterisk-users] Impossible to use any recent asterisk version with chan_sip
This is a great news, thank you. I have open the issue, https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the relevant files, let me know if you need more info. Leandro 2016-07-06 21:46 GMT+02:00 Joshua Colp <jcolp at digium.com>:> Leandro Dardini wrote: > >> Hello, >> I'd like to know if anyone of you is finding my same problems using any >> recent asterisk version, after 13.7 / 13.8 with chan_sip. >> >> If I use any recent asterisk version, after just few seconds asterisk >> completely locks up, stopping processing SIP/UDP packets. Nothing is >> written in the asterisk log, but if I run "netstat -nap | grep 5060" I >> see the UDP buffer filled up. >> >> If I step back to asterisk 13.2, then all is fine and asterisk is rock >> solid. >> >> I know I should use PJSIP and chan_sip is no more supported, but at this >> point, if this is the working state of chan_sip, it should be completely >> removed. >> > > I've responded on the issue but the backtrace you've provided makes it > appear as though the issue is actually in ODBC, which since chan_sip is > using it in your deployment it causes it to lock up (why exactly is > unknown). > > Since it's separate you should create a new issue. If you don't want to I > can do so tomorrow. The complete console output (with debug going to > console in logger.conf and core set debug 3) as well as the configuration > would also be useful. > > Cheers, > > > -- > 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160706/e6036b11/attachment.html>