Re: [asterisk-users] Advices when Asterisk segfaults and nothing useful in logs
On Tue, Feb 14, 2017 at 2:51 PM, George Joseph wrote: > > > On Tue, Feb 14, 2017 at 10:21 AM, Olivier wrote: > >> Hello, >> >> I've got a 13.13.1 system using PJSIP stack on debian Jessie. >> It runs from 50 to 100 simultaneous calls (so 100 to 200 PJSIP channels) >> all day long. >> From time to time, roughly meaning once a month, it segfaults with lines >> (from dmesg -T output) like this: >> asterisk[1160]: segfault at 7efe ip 005881d6 sp >> 7fec95c33910 error 4 in asterisk[40+2a2000] >> >> >> Debug level was unfortunately not set in asterisk.conf but verbose level >> was set to 5. >> Asterisk runs with: >> /usr/sbin/asterisk -U asterisk -G asterisk -g >> >> Asterisk is compiled with DONT_OPTIMIZE and BETTER_BACKTRACES options. >> >> "core show settings" outputs: >> * Directories >> - >> Configuration file: >> Configuration directory: /etc/asterisk >> Module directory:/usr/lib/asterisk/modules >> Spool directory: /var/spool/asterisk >> Log directory: /var/log/asterisk >> Run/Sockets directory: /var/run/asterisk >> PID file:/var/run/asterisk/asterisk.pid >> VarLib directory:/var/lib/asterisk >> Data directory: /var/lib/asterisk >> ASTDB: /var/lib/asterisk/astdb >> IAX2 Keys directory: /var/lib/asterisk/keys >> >> >> >> 1. Am I correct to expect a coredump file to be produced anytime asterisk >> segfaults ? >> > > Yes if -g is set and the user that's running asterisk has permissions to > set ulimit -c. > > >> >> 2. Does Asterisk prints any WARNING or ERROR message whenever it detects, >> at startup preferably, that it has not required permissions to write a >> coredump file ? >> > > No because it's the system that determines where a coredump goes and > actually writes it, not asterisk. > It's the sysctl kernel.core_pattern setting. > > >> >> 3. Among above directories, which one is choosen to save coredump files ? >> Is it something that can/should be configured in /etc/asterisk (I've seen >> related options in some debian /etc/default/asterisk files but I would be >> curious to know if such things exist >> > > See above. > > >> >> 4. Is there anything useful I can do with a line such as : >> asterisk[1160]: segfault at 7efe ip 005881d6 sp >> 7fec95c33910 error 4 in asterisk[40+2a2000] ? Any pointer ? >> > > Nope. Not a thing. Sorry. > > > >> >> 5. Suggestions ? >> > > If you can at least get the system to write a coredump file, there are new > utilities in /var/lib/asterisk/scripts, namely ast_coredumper which can > help create the backtraces if it can at least find the core file. Just run > "./ast_coredumper --help" for more info. You should also be able to use > those utilities with earlier Asterisk 13 versions. > > > Oh yeah, and it's on my list to publish instructions on how ot use those utilities but they were just released yesterday. > >> Best regards >> >> >> -- >> _ >> -- 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 >> > > > > -- > George Joseph > Digium, Inc. | Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- George Joseph Digium, Inc. | 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
Re: [asterisk-users] Advices when Asterisk segfaults and nothing useful in logs
On Tue, Feb 14, 2017 at 10:21 AM, Olivier wrote: > Hello, > > I've got a 13.13.1 system using PJSIP stack on debian Jessie. > It runs from 50 to 100 simultaneous calls (so 100 to 200 PJSIP channels) > all day long. > From time to time, roughly meaning once a month, it segfaults with lines > (from dmesg -T output) like this: > asterisk[1160]: segfault at 7efe ip 005881d6 sp > 7fec95c33910 error 4 in asterisk[40+2a2000] > > > Debug level was unfortunately not set in asterisk.conf but verbose level > was set to 5. > Asterisk runs with: > /usr/sbin/asterisk -U asterisk -G asterisk -g > > Asterisk is compiled with DONT_OPTIMIZE and BETTER_BACKTRACES options. > > "core show settings" outputs: > * Directories > - > Configuration file: > Configuration directory: /etc/asterisk > Module directory:/usr/lib/asterisk/modules > Spool directory: /var/spool/asterisk > Log directory: /var/log/asterisk > Run/Sockets directory: /var/run/asterisk > PID file:/var/run/asterisk/asterisk.pid > VarLib directory:/var/lib/asterisk > Data directory: /var/lib/asterisk > ASTDB: /var/lib/asterisk/astdb > IAX2 Keys directory: /var/lib/asterisk/keys > > > > 1. Am I correct to expect a coredump file to be produced anytime asterisk > segfaults ? > Yes if -g is set and the user that's running asterisk has permissions to set ulimit -c. > > 2. Does Asterisk prints any WARNING or ERROR message whenever it detects, > at startup preferably, that it has not required permissions to write a > coredump file ? > No because it's the system that determines where a coredump goes and actually writes it, not asterisk. It's the sysctl kernel.core_pattern setting. > > 3. Among above directories, which one is choosen to save coredump files ? > Is it something that can/should be configured in /etc/asterisk (I've seen > related options in some debian /etc/default/asterisk files but I would be > curious to know if such things exist > See above. > > 4. Is there anything useful I can do with a line such as : > asterisk[1160]: segfault at 7efe ip 005881d6 sp > 7fec95c33910 error 4 in asterisk[40+2a2000] ? Any pointer ? > Nope. Not a thing. Sorry. > > 5. Suggestions ? > If you can at least get the system to write a coredump file, there are new utilities in /var/lib/asterisk/scripts, namely ast_coredumper which can help create the backtraces if it can at least find the core file. Just run "./ast_coredumper --help" for more info. You should also be able to use those utilities with earlier Asterisk 13 versions. > > Best regards > > > -- > _ > -- 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 > -- George Joseph Digium, Inc. | 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
Re: [asterisk-users] Execution of pre-bridge handlers
What an excellent response Richard!!! Thank you very much for that!! Best regards! Patrick On Wed, Feb 15, 2017 at 5:23 AM, Richard Mudgett wrote: > > > On Tue, Feb 14, 2017 at 6:24 AM, Patrick Wakano wrote: > >> Hello Asterisk Users, >> >> Hope you all doing fine! >> I am working with a quite complex dialplan, and I've come to some >> situations where it makes some nasty use of pre-bridge handlers. >> The pre-bridge handlers wiki (https://wiki.asterisk.org/wik >> i/display/AST/Pre-Bridge+Handlers) doesn't have the big warning the >> pre-dial one has indicating it must return and must not put the >> caller/callee in other applications (https://wiki.asterisk.org/wik >> i/display/AST/Pre-Dial+Handlers). So apparently, looks like they >> wouldn't have this restriction... However I had the feeling this was not >> true, so after some research I found this issue >> https://issues.asterisk.org/jira/browse/ASTERISK-25690, that says "*Connected >> line subroutines are meant** to be fast and as a result there is an >> expectation that applications invoked will not consume frames*". I am >> assuming that connected lines subroutines are just different words for >> pre-bridge handlers, right? >> Anyway my question is, what happens if I do not return straight away from >> the pre-bridge handler? Or even worst, if I execute a Dial application for >> example? Will I fall in some "undefined behaviour"? >> >> Anyone has experienced something like this? >> > > There are several handler routines available and each handles situations > for the > different states of a call. It makes no sense to use the Hangup() > application in > any of them and you must return from all of them. Most of the handlers > operate > from within the Dial application. > > Pre-dial handlers > The purpose of these routines is to setup a channel to place a call. > The pre-dial > routines can be run on the calling and called channels. See the Dial > application > documentation. > > For the calling channel, you can do most anything to the calling channel > except > hangup because you are still within the Dial application's control. The > reason > for the ability to execute a pre-dial routine on the calling channel > instead of doing > all the setup before executing Dial is to eliminate a window of > opportunity when using > the Lock/Unlock applications with Dial. > > For the called channel, you can only setup the channel. At this point, > the channel > exists but is not connected to anything nor has the call been placed. > Do your > channel setup and return. > > Redirecting interception handlers > This routine normally executes on the calling channel because the called > channel > has indicated that the call is being diverted/forwarded/redirected to > somewhere > else. The purpose of this routine is to get the REDIRECTING party > information > setup as you want and then return. You do not have control of the media > nor should > you hangup. You also should be quick about it. > > Pre-bridge handlers > At this point the called channel has answered and all other called > channels that were > dialed have been hung up. The called channel is about to be bridged > with the calling > channel. > > The purpose of this routine is to give the called person an opportunity > to decide if > he even wants to talk to the caller. You have control of the media > stream to the called > party. You cannot hangup the channel in the routine because you must > return. If you > want to abort bridging the call with the channel you must set a return > value as > documented by the Dial application. You need to remember that the > caller is > waiting to be connected the entire time you are in this routine. > > Connected-line interception handlers > At this point the channels are bridged together and may have been > talking for awhile. > > The purpose of this routine is to get the CONNECTEDLINE party > information setup > as you want and then return. The bridged peer has changed identity > likely because > of a transfer. You do not have control of the media nor should you > hangup. You also > need be quick about it or you risk causing a noticeable interruption to > the media. > > Hangup handlers > At this point the channel is hungup and you should be gathering > information about > the call for further processing later. You should not be doing > extensive post call > analysis at this time because you are delaying the channel technology > hangup > sequence. You have the same restrictions with the h extension. > > Given what I have stated about pre-bridge handlers you should be able to > see that > doing a Dial in a pre-bridge handler (or any handler for that matter) is a > bad thing to > do and definitely falls into the "undefined behavior" category. > > Richard > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-di
[asterisk-users] T.38 negotiation timeout
Hello Asterisk Users. I have an issue with receiving fax on my Asterisk/SIP channel. I keep getting timeout under T.38 negotiation - see http://pastebin.com/6eCe26YM Any help would be greatly appreciated. /Jacob -- _ -- 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] Execution of pre-bridge handlers
On Tue, Feb 14, 2017 at 6:24 AM, Patrick Wakano wrote: > Hello Asterisk Users, > > Hope you all doing fine! > I am working with a quite complex dialplan, and I've come to some > situations where it makes some nasty use of pre-bridge handlers. > The pre-bridge handlers wiki (https://wiki.asterisk.org/wik > i/display/AST/Pre-Bridge+Handlers) doesn't have the big warning the > pre-dial one has indicating it must return and must not put the > caller/callee in other applications (https://wiki.asterisk.org/wik > i/display/AST/Pre-Dial+Handlers). So apparently, looks like they wouldn't > have this restriction... However I had the feeling this was not true, so > after some research I found this issue https://issues.asterisk.org/ji > ra/browse/ASTERISK-25690, that says "*Connected line subroutines are > meant** to be fast and as a result there is an expectation that > applications invoked will not consume frames*". I am assuming that > connected lines subroutines are just different words for pre-bridge > handlers, right? > Anyway my question is, what happens if I do not return straight away from > the pre-bridge handler? Or even worst, if I execute a Dial application for > example? Will I fall in some "undefined behaviour"? > > Anyone has experienced something like this? > There are several handler routines available and each handles situations for the different states of a call. It makes no sense to use the Hangup() application in any of them and you must return from all of them. Most of the handlers operate from within the Dial application. Pre-dial handlers The purpose of these routines is to setup a channel to place a call. The pre-dial routines can be run on the calling and called channels. See the Dial application documentation. For the calling channel, you can do most anything to the calling channel except hangup because you are still within the Dial application's control. The reason for the ability to execute a pre-dial routine on the calling channel instead of doing all the setup before executing Dial is to eliminate a window of opportunity when using the Lock/Unlock applications with Dial. For the called channel, you can only setup the channel. At this point, the channel exists but is not connected to anything nor has the call been placed. Do your channel setup and return. Redirecting interception handlers This routine normally executes on the calling channel because the called channel has indicated that the call is being diverted/forwarded/redirected to somewhere else. The purpose of this routine is to get the REDIRECTING party information setup as you want and then return. You do not have control of the media nor should you hangup. You also should be quick about it. Pre-bridge handlers At this point the called channel has answered and all other called channels that were dialed have been hung up. The called channel is about to be bridged with the calling channel. The purpose of this routine is to give the called person an opportunity to decide if he even wants to talk to the caller. You have control of the media stream to the called party. You cannot hangup the channel in the routine because you must return. If you want to abort bridging the call with the channel you must set a return value as documented by the Dial application. You need to remember that the caller is waiting to be connected the entire time you are in this routine. Connected-line interception handlers At this point the channels are bridged together and may have been talking for awhile. The purpose of this routine is to get the CONNECTEDLINE party information setup as you want and then return. The bridged peer has changed identity likely because of a transfer. You do not have control of the media nor should you hangup. You also need be quick about it or you risk causing a noticeable interruption to the media. Hangup handlers At this point the channel is hungup and you should be gathering information about the call for further processing later. You should not be doing extensive post call analysis at this time because you are delaying the channel technology hangup sequence. You have the same restrictions with the h extension. Given what I have stated about pre-bridge handlers you should be able to see that doing a Dial in a pre-bridge handler (or any handler for that matter) is a bad thing to do and definitely falls into the "undefined behavior" category. Richard -- _ -- 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] Advices when Asterisk segfaults and nothing useful in logs
Hello, I've got a 13.13.1 system using PJSIP stack on debian Jessie. It runs from 50 to 100 simultaneous calls (so 100 to 200 PJSIP channels) all day long. >From time to time, roughly meaning once a month, it segfaults with lines (from dmesg -T output) like this: asterisk[1160]: segfault at 7efe ip 005881d6 sp 7fec95c33910 error 4 in asterisk[40+2a2000] Debug level was unfortunately not set in asterisk.conf but verbose level was set to 5. Asterisk runs with: /usr/sbin/asterisk -U asterisk -G asterisk -g Asterisk is compiled with DONT_OPTIMIZE and BETTER_BACKTRACES options. "core show settings" outputs: * Directories - Configuration file: Configuration directory: /etc/asterisk Module directory:/usr/lib/asterisk/modules Spool directory: /var/spool/asterisk Log directory: /var/log/asterisk Run/Sockets directory: /var/run/asterisk PID file:/var/run/asterisk/asterisk.pid VarLib directory:/var/lib/asterisk Data directory: /var/lib/asterisk ASTDB: /var/lib/asterisk/astdb IAX2 Keys directory: /var/lib/asterisk/keys 1. Am I correct to expect a coredump file to be produced anytime asterisk segfaults ? 2. Does Asterisk prints any WARNING or ERROR message whenever it detects, at startup preferably, that it has not required permissions to write a coredump file ? 3. Among above directories, which one is choosen to save coredump files ? Is it something that can/should be configured in /etc/asterisk (I've seen related options in some debian /etc/default/asterisk files but I would be curious to know if such things exist 4. Is there anything useful I can do with a line such as : asterisk[1160]: segfault at 7efe ip 005881d6 sp 7fec95c33910 error 4 in asterisk[40+2a2000] ? Any pointer ? 5. Suggestions ? Best regards -- _ -- 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] 14.3.0 download archive corrupt - cannot extract
is back online now thanks! On Feb 14, 2017, 11:18 -0300, Joshua Colp , wrote: > On Tue, Feb 14, 2017, at 10:13 AM, Sebastian Gutierrez wrote: > > The 13.14 tar gz doesn’t even exists on the current or in the old > > releases folder. > > > > there seems to be an issue with the latest build not generating the > > artifacts? > > It was temporarily removed during a synchronization but is now back up > and the issue should be resolved. > > -- > 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 -- _ -- 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] 14.3.0 download archive corrupt - cannot extract
On Tue, Feb 14, 2017, at 10:13 AM, Sebastian Gutierrez wrote: > The 13.14 tar gz doesn’t even exists on the current or in the old > releases folder. > > there seems to be an issue with the latest build not generating the > artifacts? It was temporarily removed during a synchronization but is now back up and the issue should be resolved. -- 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
Re: [asterisk-users] 14.3.0 download archive corrupt - cannot extract
The 13.14 tar gz doesn’t even exists on the current or in the old releases folder. there seems to be an issue with the latest build not generating the artifacts? best regards On Feb 14, 2017, 11:04 -0300, Marcelo Terres , wrote: > Thanks Joshua. > Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres > https://linkedin.com/in/marceloterres > > > On 14 February 2017 at 14:01, Joshua Colp wrote: > > On Tue, Feb 14, 2017, at 09:57 AM, Marcelo Terres wrote: > > > Same problem with me. > > > > > > I downloaded the file in 2 different places and had the same error... > > > > An issue was filed for tracking this[1] and it will be resolved later > > today. > > > > [1] https://issues.asterisk.org/jira/browse/ASTERISK-26791 > > > > -- > > 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 > > -- > _ > -- 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 -- _ -- 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] 14.3.0 download archive corrupt - cannot extract
Thanks Joshua. Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 14 February 2017 at 14:01, Joshua Colp wrote: > On Tue, Feb 14, 2017, at 09:57 AM, Marcelo Terres wrote: >> Same problem with me. >> >> I downloaded the file in 2 different places and had the same error... > > An issue was filed for tracking this[1] and it will be resolved later > today. > > [1] https://issues.asterisk.org/jira/browse/ASTERISK-26791 > > -- > 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 -- _ -- 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] 14.3.0 download archive corrupt - cannot extract
On Tue, Feb 14, 2017, at 09:57 AM, Marcelo Terres wrote: > Same problem with me. > > I downloaded the file in 2 different places and had the same error... An issue was filed for tracking this[1] and it will be resolved later today. [1] https://issues.asterisk.org/jira/browse/ASTERISK-26791 -- 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
Re: [asterisk-users] 14.3.0 download archive corrupt - cannot extract
Same problem with me. I downloaded the file in 2 different places and had the same error... Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 14 February 2017 at 08:42, Jonathan H wrote: > Hi there; > > 2 linux boxes and Windows all report an error and the archive is not > extractable. > > Wget reports the size as follows: > > 2017-02-14 08:36:21 (7.29 MB/s) - ‘asterisk-14-current.tar.gz’ saved > [40653605/40653605] > > It starts un-tarring but then > > asterisk-14.3.0/bridges/bridge_native_rtp.c > asterisk-14.3.0/sounds/ > asterisk-14.3.0/sounds/asterisk-core-sounds-en-gsm-1.5.tar.gz > > gzip: stdin: invalid compressed data--format violated > tar: Unexpected EOF in archive > tar: Unexpected EOF in archive > tar: Error is not recoverable: exiting now > > > > > > > > -- > _ > -- 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 -- _ -- 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] Execution of pre-bridge handlers
Hello Asterisk Users, Hope you all doing fine! I am working with a quite complex dialplan, and I've come to some situations where it makes some nasty use of pre-bridge handlers. The pre-bridge handlers wiki (https://wiki.asterisk.org/ wiki/display/AST/Pre-Bridge+Handlers) doesn't have the big warning the pre-dial one has indicating it must return and must not put the caller/callee in other applications (https://wiki.asterisk.org/ wiki/display/AST/Pre-Dial+Handlers). So apparently, looks like they wouldn't have this restriction... However I had the feeling this was not true, so after some research I found this issue https://issues.asterisk.org/ jira/browse/ASTERISK-25690, that says "*Connected line subroutines are meant** to be fast and as a result there is an expectation that applications invoked will not consume frames*". I am assuming that connected lines subroutines are just different words for pre-bridge handlers, right? Anyway my question is, what happens if I do not return straight away from the pre-bridge handler? Or even worst, if I execute a Dial application for example? Will I fall in some "undefined behaviour"? Anyone has experienced something like this? Many thanks, Cheers, Patrick -- _ -- 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] 14.3.0 download archive corrupt - cannot extract
Hi there; 2 linux boxes and Windows all report an error and the archive is not extractable. Wget reports the size as follows: 2017-02-14 08:36:21 (7.29 MB/s) - ‘asterisk-14-current.tar.gz’ saved [40653605/40653605] It starts un-tarring but then asterisk-14.3.0/bridges/bridge_native_rtp.c asterisk-14.3.0/sounds/ asterisk-14.3.0/sounds/asterisk-core-sounds-en-gsm-1.5.tar.gz gzip: stdin: invalid compressed data--format violated tar: Unexpected EOF in archive tar: Unexpected EOF in archive tar: Error is not recoverable: exiting now -- _ -- 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
