Re: [asterisk-users] Segmentation fault
Hi Federico, The first hit from Google 'how to run command from gdb' https://ftp.gnu.org/old-gnu/Manuals/gdb/html_chapter/gdb_5.html#:~:text=Use%20the%20run%20command%20to,section%20Commands%20to%20specify%20files). # gdb (gdb) file /usr/sbin/asterisk (gdb) run –gvvc Starting program: /usr/sbin/asterisk -gvc [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". snip Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. snip... etc etc On 8/20/23 09:12, Federico wrote: I cannot follow your instructions, because asterisk segfaults on start. It never starts Can you give me instruction to trap this segfault on starting asterisk? Like gdb …..asterist –gvvc *From:* asterisk-users *On Behalf Of *Mark Murawski *Sent:* Saturday, August 19, 2023 11:04 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] Segmentation fault Hi Federico, Segfaults are 100% not by design. Typically if something seg faulted, either there is a logic bug or a component mismatch. The you should definitely be able to use more than one connection (we use multiple connections with postgres odbc with no issue). If Asterisk segfaults when using odbc Try this: - use ps and get the pid of Asterisk - run gdb, attach to the asterisk pid - do something that would cause the seg fault - get a backtrace (bt) and show all threads backtrace (thread apply all bt) if Asterisk segfaults when starting up Run Asterisk straight from gdb Wait for segfault, get backtrace, and all threads backtrace On 8/16/23 18:48, Federico wrote: I tested this issue with version 13 and version 18. In res_odbc.conf, if I add a second, new data source like [asterisk] enabled=yes dsn=asterisk sanitysql => select 1 isolation => read_committed username=root ;password= pre-connect => yes forcecommit => yes connect_timeout => 10 negative_connection_cache => 0 max_connections =>500 my odbc.ini [cdr] Description = MySQL ODBC Driver Testing Driver = maria Socket = /var/run/mysqld/mysqld.sock User = root Password = Database = public Option = 3 I get, immediately, segmentation fault. With only one, it works fine. Is this by design? Philip -- _ -- 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] Segmentation fault
I cannot follow your instructions, because asterisk segfaults on start. It never starts Can you give me instruction to trap this segfault on starting asterisk? Like gdb …..asterist –gvvc From: asterisk-users On Behalf Of Mark Murawski Sent: Saturday, August 19, 2023 11:04 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Segmentation fault Hi Federico, Segfaults are 100% not by design. Typically if something seg faulted, either there is a logic bug or a component mismatch. The you should definitely be able to use more than one connection (we use multiple connections with postgres odbc with no issue). If Asterisk segfaults when using odbc Try this: - use ps and get the pid of Asterisk - run gdb, attach to the asterisk pid - do something that would cause the seg fault - get a backtrace (bt) and show all threads backtrace (thread apply all bt) if Asterisk segfaults when starting up Run Asterisk straight from gdb Wait for segfault, get backtrace, and all threads backtrace On 8/16/23 18:48, Federico wrote: I tested this issue with version 13 and version 18. In res_odbc.conf, if I add a second, new data source like [asterisk] enabled=yes dsn=asterisk sanitysql => select 1 isolation => read_committed username=root ;password= pre-connect => yes forcecommit => yes connect_timeout => 10 negative_connection_cache => 0 max_connections =>500 my odbc.ini [cdr] Description = MySQL ODBC Driver Testing Driver = maria Socket = /var/run/mysqld/mysqld.sock User = root Password = Database = public Option = 3 I get, immediately, segmentation fault. With only one, it works fine. Is this by design? Philip -- _ -- 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] Segmentation fault
Hi Federico, Segfaults are 100% not by design. Typically if something seg faulted, either there is a logic bug or a component mismatch. The you should definitely be able to use more than one connection (we use multiple connections with postgres odbc with no issue). If Asterisk segfaults when using odbc Try this: - use ps and get the pid of Asterisk - run gdb, attach to the asterisk pid - do something that would cause the seg fault - get a backtrace (bt) and show all threads backtrace (thread apply all bt) if Asterisk segfaults when starting up Run Asterisk straight from gdb Wait for segfault, get backtrace, and all threads backtrace On 8/16/23 18:48, Federico wrote: I tested this issue with version 13 and version 18. In res_odbc.conf, if I add a second, new data source like [asterisk] enabled=yes dsn=asterisk sanitysql => select 1 isolation => read_committed username=root ;password= pre-connect => yes forcecommit => yes connect_timeout => 10 negative_connection_cache => 0 max_connections =>500 my odbc.ini [cdr] Description = MySQL ODBC Driver Testing Driver = maria Socket = /var/run/mysqld/mysqld.sock User = root Password = Database = public Option = 3 I get, immediately, segmentation fault. With only one, it works fine. Is this by design? Philip -- _ -- 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] Segmentation fault
On Wed, Aug 16, 2023 at 7:48 PM Federico wrote: > I tested this issue with version 13 and version 18. > > In res_odbc.conf, if I add a second, new data source like > > > > [asterisk] > > enabled=yes > > dsn=asterisk > > sanitysql => select 1 > > isolation => read_committed > > username=root > > ;password= > > pre-connect => yes > > forcecommit => yes > > connect_timeout => 10 > > negative_connection_cache => 0 > > max_connections =>500 > > > > my odbc.ini > > [cdr] > > Description = MySQL ODBC Driver Testing > > Driver = maria > > Socket = /var/run/mysqld/mysqld.sock > > User = root > > Password = > > Database = public > > Option = 3 > > > > > > I get, immediately, segmentation fault. > > With only one, it works fine. > > Is this by design? > There's nothing explicitly written to prevent such a thing within Asterisk itself. There is no backtrace here so nothing to show where the crash actually occurred, be it Asterisk itself or UnixODBC. If UnixODBC we're a somewhat simple user of it, so specific configuration of it or its build may be the source of it. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and 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] Segmentation fault
>Setting "batch=yes" in your /etc/asterisk/cdr.conf file may fix this by moving the database interaction to separate threads away from the call. I can certainly try this, but I believe the maria-odbc-driver is what is running as a single thread. Its been a few months since I played with this, I think there may also be a compiled in parameter in the unixodbc regarding threading that's default value was changed in recent years. The only asterisk issue is why it would hang at SET(CDR_PROP(disable)=1)? Perhaps there is an interaction where odbc_adaptive always checks an aspect of the tables it interacts with (and needlessly when disabling cdr). I started looking at the source - but got pulled onto another project before I was able to determine anything, and have not revisited the issue. So for me this isn't current but just me throwing in my recollections that may pertain to Federico's issue. On Thu, Aug 17, 2023 at 5:37 PM C. Maj wrote: > On 8/17/23 12:44, John Harragin wrote: > > You should be able to define multiple data sources. However I'm having my > > own issues. I have my dialplan accessing one maria database which is > hosted > > locally on the asterisk server then logging cdr with odbc adaptive which > > connects to maria on a remote machine. This works fine except when the > > remote server is out of reach the calls zombie and eventually lead to a > > fault. > > Setting "batch=yes" in your /etc/asterisk/cdr.conf file may fix this by > moving the database interaction to separate threads away from the call. > > -- > _ > -- 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] Segmentation fault
On 8/17/23 12:44, John Harragin wrote: You should be able to define multiple data sources. However I'm having my own issues. I have my dialplan accessing one maria database which is hosted locally on the asterisk server then logging cdr with odbc adaptive which connects to maria on a remote machine. This works fine except when the remote server is out of reach the calls zombie and eventually lead to a fault. Setting "batch=yes" in your /etc/asterisk/cdr.conf file may fix this by moving the database interaction to separate threads away from the call. -- _ -- 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] Segmentation fault
You should be able to define multiple data sources. However I'm having my own issues. I have my dialplan accessing one maria database which is hosted locally on the asterisk server then logging cdr with odbc adaptive which connects to maria on a remote machine. This works fine except when the remote server is out of reach the calls zombie and eventually lead to a fault. The cli imply that the calls hang on SET(CDR_PROP(disable)=1) which is ironic since instructing the system not to connect to the cdr server appears to cause the problem and there may be a defect in the adaptive_odbc implementation where it somehow still communicates with the server during the cdr_func(disable). My experimentation suggests that the maria_odbc driver utilizes a single thread (there are odbc directives to adjust this - but it seems they have been superseded). Once a cdr hangs, new calls hang on odbc calls to the local database. In my case the remote server is 5' away and in the same switch that Asterisk is plugged into, so it hasn't been critical to fix this - but it may provide some clues to your situation. On Thu, Aug 17, 2023 at 11:04 AM Federico wrote: > I tested this issue with version 13 and version 18. > > In res_odbc.conf, if I add a second, new data source like > > > > [asterisk] > > enabled=yes > > dsn=asterisk > > sanitysql => select 1 > > isolation => read_committed > > username=root > > ;password= > > pre-connect => yes > > forcecommit => yes > > connect_timeout => 10 > > negative_connection_cache => 0 > > max_connections =>500 > > > > my odbc.ini > > [cdr] > > Description = MySQL ODBC Driver Testing > > Driver = maria > > Socket = /var/run/mysqld/mysqld.sock > > User = root > > Password = > > Database = public > > Option = 3 > > > > > > I get, immediately, segmentation fault. > > With only one, it works fine. > > Is this by design? > > > > Philip > > > > > > > -- > _ > -- 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] Segmentation fault
I tested this issue with version 13 and version 18. In res_odbc.conf, if I add a second, new data source like [asterisk] enabled=yes dsn=asterisk sanitysql => select 1 isolation => read_committed username=root ;password= pre-connect => yes forcecommit => yes connect_timeout => 10 negative_connection_cache => 0 max_connections =>500 my odbc.ini [cdr] Description = MySQL ODBC Driver Testing Driver = maria Socket = /var/run/mysqld/mysqld.sock User = root Password = Database = public Option = 3 I get, immediately, segmentation fault. With only one, it works fine. Is this by design? Philip -- _ -- 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] Segmentation Fault Asterisk 13.7.0-rc2 (libmysqlclient?)
I compiled Asterisk 13.5.0 (and 13.7.0-rc2), PJproject 2.4.5, and DAHDI 2.11.0 from source. I am using Debian 8.2 and mysql is version 5.5.46. This is a recent problem and when we went back to 13.5.0 and started seeing the issue, we started thinking that mysql must have updated and caused the problem. I think you are spot-on with your analysis. I will do a little experimentation with mysql and see if I can isolate where the problem started. Thanks, --Matt From: asterisk-users-boun...@lists.digium.com on behalf of A J Stiles Sent: Tuesday, January 19, 2016 7:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Segmentation Fault Asterisk 13.7.0-rc2 (libmysqlclient?) On Monday 18 Jan 2016, Matthew Murphy wrote: > Hi everyone, > > I am getting a segmentation fault (seems to occur randomly) using Asterisk > 13.7.0-rc2 with PJProject 2.4.5. It appears to be something that > libmysqlclient is complaining about when doing a query in > ps_endpoint_id_ips. We are using Asterisk Realtime. This also seems to > occur in Asterisk 13.5.0. Which bits did you compile from Source Code yourself, and which bits (if any) are precompiled by your distribution? What is your libmysqlclient version? If you are using some sort of Ubuntu / Debian-based distribution, could it possibly have done a sneaky `apt-get upgrade` behind your back? -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Segmentation Fault Asterisk 13.7.0-rc2 (libmysqlclient?)
On Monday 18 Jan 2016, Matthew Murphy wrote: > Hi everyone, > > I am getting a segmentation fault (seems to occur randomly) using Asterisk > 13.7.0-rc2 with PJProject 2.4.5. It appears to be something that > libmysqlclient is complaining about when doing a query in > ps_endpoint_id_ips. We are using Asterisk Realtime. This also seems to > occur in Asterisk 13.5.0. Which bits did you compile from Source Code yourself, and which bits (if any) are precompiled by your distribution? What is your libmysqlclient version? If you are using some sort of Ubuntu / Debian-based distribution, could it possibly have done a sneaky `apt-get upgrade` behind your back? -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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
[asterisk-users] Segmentation Fault Asterisk 13.7.0-rc2 (libmysqlclient?)
Hi everyone, I am getting a segmentation fault (seems to occur randomly) using Asterisk 13.7.0-rc2 with PJProject 2.4.5. It appears to be something that libmysqlclient is complaining about when doing a query in ps_endpoint_id_ips. We are using Asterisk Realtime. This also seems to occur in Asterisk 13.5.0. Below is a backtrace that might help a little. I have looked through the change log for the 13.7.0 release and some of items addressed may fix my problem. Before diving in and attempting to upgrade to the final version of 13.7.0, I was hoping someone with knowledge would be able to look at this and let me know if this is something already seen or if this is entirely new. Thanks for the help! --Matt --- BACKTRACE BELOW --- [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `asterisk -g'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x7f1e02e8a120 in list_add () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.18 #0 0x7f1e02e8a120 in list_add () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.18 No symbol table info available. #1 0x7f1e0339d132 in my_SQLAllocStmt () from /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so No symbol table info available. #2 0x7f1e38354af4 in ?? () from /usr/lib/x86_64-linux-gnu/libodbc.so.2 No symbol table info available. #3 0x7f1e01e8f16f in custom_prepare (obj=0x1e588c8, data=0x7f1dc8a289b0) at res_config_odbc.c:107 res = 0 x = 1 count = 0 cps = 0x7f1dc8a289b0 field = 0x1e588a8 encodebuf = "\001\200\255\373\000\000\000\001P\205\242\310\035\177\000\000P\205\242\310\035\177\000\000P\205\242\310\035\177\000\000P\205\242\310\035\177\000\000\206\205\242\310\035\177\000\000\237\205\242\310\035\177\000\000P\205\242\310\035\177\000\000\237\205\242\310\035\177\000\000\342X\350\002\036\177\000\000\247p\350\001\000\000\000\000\351ʎ:\036\177\000\000\000\000\000\000\000\000\000\000\335Y\350\002\036\177\000\000`\000\000\000\004\000\000\000\a\000\000\000\000\000\000\000@\256\350\001\000\000\000\000\a", '\000' , "\340U\347\002\036\177\000\000\a\000\000\000\000\000\000\000;\000\000\000\000\000\000\000\002\000\000\000\035\177\000\001`\204\242\310\035\177\000\000\377\377\377\377\000\000\000\000\000"... stmt = 0x0 __PRETTY_FUNCTION__ = "custom_prepare" #4 0x7f1e385b9783 in ast_odbc_prepare_and_execute (obj=0x1e588c8, prepare_cb=0x7f1e01e8f11f , data=0x7f1dc8a289b0) at res_odbc.c:640 res = 0 i = 0 attempt = 0 nativeerror = 0 numfields = 0 diagbytes = 0 state = "\v:\351\001\036\177\000\000t\001" diagnostic = "\340\030\006\244\035\177\000\000̜\242\310\035\177\000\000\200\020\024\002\000\000\000\000,Z_9\036\177\000\000)\266\227V\000\000\000\000\320\003\000\000\000\000\000\000\240\210\242\310\035\177\000\000\350\341\000D\035\177\000\000\300s\276\314\035\177\000\000\255I^\000\000\000\000\000\350\341\000D\035\177\000\000p>d\000\000\000\000\000̜\242\310\034\b\000\000K6d\000\000\000\000\000\320\003\000\000\000\000\000\000\001\000\000\000\000\000\000\000`\211\242\310\035\177\000\000Г\002\244\035\177\000\000\360\210\242\310\035\177\000\000\065\200^\000\000\000\000\000@F\351\001\036\177\000\000`\211\242\310t\001\000\000\v:\351\001\036\177\000\000\000\001\000\000\000\000\000\000\300\211\242\310\035\177\000\000ȑ\242\310"... stmt = 0x7f1dc8a297b0 __PRETTY_FUNCTION__ = "ast_odbc_prepare_and_execute" #5 0x7f1e01e90715 in realtime_multi_odbc (database=0x7f1dc8a298d0 "asterisk", table=0x7f1dc8a297d0 "ps_endpoint_id_ips", fields=0x7f1da40185f0) at res_config_odbc.c:376 obj = 0x1e588c8 stmt = 0x25ed6b2 sql = "SELECT * FROM ps_endpoint_id_ips WHERE id LIKE ? ORDER BY id\000\177\000\000\377\377\377\377\377\377\377\377\000\000\000\000\000\000\000\000`\224\242\310\035\177\000\000m\225\242\310\035\177\000\000`\225\242\310\035\177\000\000`\226\242\310\035\177\000\000@\225\242\310\035\177\000\000\033\311d9\036\177\000\000P\225\242\310\035\177\000\000ȕ\242\310", '\000' , "\377\377\377\377", '\000' , " R\234\317\035\177\000\000\000\000\000\000\035\177", '\000' ... coltitle = "\002", '\000' , "|\000\000\000\000\000\000\000ǫ\b\316", '\000' , "\002", '\000' , "\021\000\000\000\000\000\000\000\035\177\000\000\000\000\000\000\000\000\000\000 \000\000\000\000\062\062\062\000\000\000\000\035\177\000\000\000\000\000\000\035\177\000\000\000\000\000\000\035\177\000\000\000\000\000\000\036\177\000\000\377\377\377\377\035\177\000\000\000\000\000\000\036\177", '\000' , "\377\377\377\377\377\377\377\377%", '\000' , "\372I\\9\036\177\000\000\000\224\242\310\035\000\000\000\062"... rowdata = 0x7f1da4001110 initfield = 0x7f1dc8a28960 "id" op = 0x7f1e01e93b2d "" field = 0x0 stri
Re: [asterisk-users] Segmentation fault with 13.5.0 / PJSIP 2.4.5
> > Information is on the wiki[1] which details how to get a backtrace from a > core dump which may be present. Otherwise it's unlikely that the problem can > be located without exact details for reproducing it. If you can get a > backtrace though you can file an issue on the issue tracker[2]. > > [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace > [2] https://issues.asterisk.org/jira > Thanks for the references. I am playing around with Asterisk several months now and never have experienced a segfault until yesterday. I always thought the software would be stable (because I have heard that even large companies use it) and haven't compiled with the options for a core dump, so it looks like that we'll never know what was happening. But I am planning to upgrade to 13.6.0 anyway, and when doing so, I'll enable the core dumps like shown in [1]. Regards, Recursive -- _ -- 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
Re: [asterisk-users] Segmentation fault with 13.5.0 / PJSIP 2.4.5
Recursive wrote: Dear colleagues, I just have experienced a segmentation fault with Asterisk 13.5.0 and PJSIP 2.4.5. Both of them have been compiled on a standard Debian Wheezy 64 bit. I did not apply any patch or alter the sources of Asterisk or PJSIP in any way. Before compiling and installing, I removed all traces of all old Asterisk and PJSIP versions from my system very thoroughly. The segmentation fault occurred when I was re-registering multiple endpoints one after another as fast as I could do it manually (by operating the respective phones) and at the same time tried to make calls from these phones. I have attached what was happening before the segfault in the CLI; I had started Asterisk with -vc. Unfortunately, I can't reproduce the situation, so if we want to know the cause of the problem, I guess it's necessary that a developer analyzes the attachment and the respective part of the source code. I think I am not (yet ?) so deep in Asterisk that I could help debug the problem, but if anybody tells me what to do I am willing to help. Information is on the wiki[1] which details how to get a backtrace from a core dump which may be present. Otherwise it's unlikely that the problem can be located without exact details for reproducing it. If you can get a backtrace though you can file an issue on the issue tracker[2]. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace [2] https://issues.asterisk.org/jira -- 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
[asterisk-users] Segmentation fault with 13.5.0 / PJSIP 2.4.5
Dear colleagues,
I just have experienced a segmentation fault with Asterisk 13.5.0 and PJSIP
2.4.5. Both of them have been compiled on a standard Debian Wheezy 64 bit. I
did not apply any patch or alter the sources of Asterisk or PJSIP in any way.
Before compiling and installing, I removed all traces of all old Asterisk and
PJSIP versions from my system very thoroughly.
The segmentation fault occurred when I was re-registering multiple endpoints
one after another as fast as I could do it manually (by operating the
respective phones) and at the same time tried to make calls from these phones.
I have attached what was happening before the segfault in the CLI; I had
started Asterisk with -vc.
Unfortunately, I can't reproduce the situation, so if we want to know the cause
of the problem, I guess it's necessary that a developer analyzes the attachment
and the respective part of the source code. I think I am not (yet ?) so deep in
Asterisk that I could help debug the problem, but if anybody tells me what to
do I am willing to help.
I hope that the attachment won't get blocked ...
Regards,
Recursive
*CLI> -- Removed contact 'sip:hfc1z...@192.168.xx.yy:5060;line=cvx8p4nk'
from AOR 'hFc1zQKs' due to request
Contact hFc1zQKs/sip:hfc1z...@192.168.xx.yy:5060;line=cvx8p4nk has been deleted
Endpoint hFc1zQKs is now Unreachable
-- Removed contact 'sip:e5jt6...@192.168.xx.yy:5060;line=7d88y4uy' from AOR
'e5JT6VqN' due to request
Contact e5JT6VqN/sip:e5jt6...@192.168.xx.yy:5060;line=7d88y4uy has been deleted
Endpoint e5JT6VqN is now Unreachable
-- Attempted to remove non-existent contact
'sip:hfc1z...@192.168.xx.yy:5060;line=cvx8p4nk' from AOR 'hFc1zQKs' by request
-- Removed contact 'sip:zyw0l...@192.168.xx.yy:5060;line=it15kl8j' from AOR
'Zyw0lsGa' due to request
Contact Zyw0lsGa/sip:zyw0l...@192.168.xx.yy:5060;line=it15kl8j has been deleted
Endpoint Zyw0lsGa is now Unreachable
-- Attempted to remove non-existent contact
'sip:e5jt6...@192.168.xx.yy:5060;line=7d88y4uy' from AOR 'e5JT6VqN' by request
-- Added contact 'sip:zyw0l...@192.168.xx.yy:5060;line=vvsqirer' to AOR
'Zyw0lsGa' with expiration of 3600 seconds
Contact Zyw0lsGa/sip:zyw0l...@192.168.xx.yy:5060;line=vvsqirer has been created
Endpoint Zyw0lsGa is now Reachable
-- Executing [0172xxx@guV4QKtp_inbound:1]
GotoIf("PJSIP/xDHoh0oP-", "0?forbidden,201") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:2]
GotoIf("PJSIP/xDHoh0oP-", "0?forbidden,201") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:3]
GotoIf("PJSIP/xDHoh0oP-", "0?forbidden,201") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:4]
GotoIf("PJSIP/xDHoh0oP-", "0?forbidden,201") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:5]
GotoIf("PJSIP/xDHoh0oP-", "0?forbidden,201") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:6]
GotoIf("PJSIP/xDHoh0oP-", "0?forbidden,201") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:7]
GotoIf("PJSIP/xDHoh0oP-", "0?101") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:8]
GotoIf("PJSIP/xDHoh0oP-", "0?101") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:9]
GotoIf("PJSIP/xDHoh0oP-", "0?101") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:10]
GotoIf("PJSIP/xDHoh0oP-", "1?101") in new stack
-- Goto (guV4QKtp_inbound,0172xxx,101)
-- Executing [0172xxx@guV4QKtp_inbound:101]
NoOp("PJSIP/xDHoh0oP-", "xDHoh0oP") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:102]
GotoIf("PJSIP/xDHoh0oP-", "0?111") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:103]
GotoIf("PJSIP/xDHoh0oP-", "0?116") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:104]
GotoIf("PJSIP/xDHoh0oP-", "0?121") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:105]
Goto("PJSIP/xDHoh0oP-", "forbidden,201") in new stack
-- Goto (guV4QKtp_inbound,forbidden,201)
-- Executing [forbidden@guV4QKtp_inbound:201]
Hangup("PJSIP/xDHoh0oP-", "21") in new stack
== Spawn extension (guV4QKtp_inbound, forbidden, 201) exited non-zero on
'PJSIP/xDHoh0oP-'
-- Added contact 'sip:hfc1z...@192.168.xx.yy:5060;line=zzizib5p' to AOR
'hFc1zQKs' with expiration of 600 seconds
Contact hFc1zQKs/sip:hfc1z...@192.168.xx.yy:5060;line=zzizib5p has been created
Endpoint hFc1zQKs is now Reachable
-- Added contact 'sip:e5jt6...@192.168.xx.yy:5060;line=ydffladj' to AOR
'e5JT6VqN' with expiration of 600 seconds
Contact e5JT6VqN/sip:e5jt6...@192.168.xx.yy:5060;line=ydffladj has been created
Endpoint e5JT6VqN is now Reachable
-- Removed contact 'sip:zyw0l...@192.168.xx.yy:5060;line=vvsqirer' from AOR
'Zyw0lsGa' due to request
Contact Zyw0lsGa/sip:zyw0l...@192.168.xx.yy:5060;line=vvsqirer has been deleted
Endpoint Zyw0lsGa is now Unreachable
-- Adde
Re: [asterisk-users] Segmentation fault after upgrading from asterisk-10.5.0 to asterisk-11.1.2
I see asterisk is finding res_jabber.so not compiled for your asterisk version. As Tim just said, remove all the modules from /usr/lib/asterisk/modules and reinstall asterisk. [2013-01-10 14:20:10] WARNING[27062]: loader.c:804 inspect_module: Module 'res_jabber.so' was not compiled with the same compile-time options as this version of Asterisk. ** [2013-01-10 14:20:10] WARNING[27062]: loader.c:805 inspect_module: Module 'res_jabber.so' will not be initialized as it may cause instability. [2013-01-10 14:20:10] WARNING[27062]: loader.c:895 load_resource: Module 'res_jabber.so' could not be loaded. Leandro 2013/1/10 Tim Nelson > First thing to *ALWAYS* check is if you have any Asterisk version specific > modules (Fax for Asterisk, G.729, etc). Ensure these are not loaded (noload > in modules.conf, or simply move them out of the asterisk modules dir). > > Tim Nelson > Systems/Network Support > Rockbochs Inc. > (218)727-4332 x105 > > -- > > After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a > Segmentation fault. > > > > [root@localhost asterisk-11.1.2]# asterisk -vvc > > Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others. > > Created by Mark Spencer > > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for > details. > > This is free software, with components licensed under the GNU General > Public > > License version 2 and other licenses; you are welcome to redistribute it > under > > certain conditions. Type 'core show license' for details. > > = > > == Parsing '/etc/asterisk/asterisk.conf': Found > > == Parsing '/etc/asterisk/extconfig.conf': Found > > == Parsing '/etc/asterisk/logger.conf': Found > > == Manager registered action DBGet > > == Manager registered action DBPut > > == Manager registered action DBDel > > == Manager registered action DBDelTree > > == Registered custom function 'MESSAGE' > > == Registered custom function 'MESSAGE_DATA' > > == Registered application 'MessageSend' > > == Manager registered action MessageSend > > == Manager registered action DataGet > > == Parsing '/etc/asterisk/codecs.conf': Found > > Asterisk Dynamic Loader Starting: > > == Parsing '/etc/asterisk/modules.conf': Found > > == Parsing '/etc/asterisk/dnsmgr.conf': Found > > [2013-01-10 14:20:10] ERROR[27062]: config_options.c:512 > aco_process_config: Unable to load config file 'acl.conf' > > == Parsing '/etc/asterisk/http.conf': Found > > == Manager registered action Ping > > == Manager registered action Events > > == Manager registered action Logoff > > == Manager registered action Login > > == Manager registered action Challenge > > == Manager registered action Hangup > > == Manager registered action Status > > == Manager registered action Setvar > > == Manager registered action Getvar > > == Manager registered action GetConfig > > == Manager registered action GetConfigJSON > > == Manager registered action UpdateConfig > > == Manager registered action CreateConfig > > == Manager registered action ListCategories > > == Manager registered action Redirect > > == Manager registered action Atxfer > > == Manager registered action Originate > > == Manager registered action Command > > == Manager registered action ExtensionState > > == Manager registered action PresenceState > > == Manager registered action AbsoluteTimeout > > == Manager registered action MailboxStatus > > == Manager registered action MailboxCount > > == Manager registered action ListCommands > > == Manager registered action SendText > > == Manager registered action UserEvent > > == Manager registered action WaitEvent > > == Manager registered action CoreSettings > > == Manager registered action CoreStatus > > == Manager registered action Reload > > == Manager registered action CoreShowChannels > > == Manager registered action ModuleLoad > > == Manager registered action ModuleCheck > > == Manager registered action AOCMessage > > == Manager registered action Filter > > == Registered custom function 'AMI_CLIENT' > > == Parsing '/etc/asterisk/manager.conf': Found > > == Parsing '/etc/asterisk/manager_humbug.conf': Found > > [2013-01-10 14:20:10] NOTICE[27062]: manager.c:7545 __init_manager: > Invalid keyword = in manager.conf [general] > > == Parsing '/etc/asterisk/users.conf': Found > > == Parsing '/etc/asterisk/cdr.conf': Found > > [2013-01-10 14:20:10] NOTICE[27062]: cdr.c:1613 do_reload: CDR logging > disabled, data will be lost. > > -- CEL logging disabled. > > == Parsing '/etc/asterisk/udptl.conf': Found > > [2013-01-10 14:20:10] WARNING[27062]: udptl.c:1413 > removed_options_handler: t38faxudpec in udptl.conf is no longer supported; > use the t38pt_udptl configuration option in sip.conf instead. > > [2013-01-10 14:20:10] WARNING[27062]: udptl.c:1415 > removed_optio
Re: [asterisk-users] Segmentation fault after upgrading from asterisk-10.5.0 to asterisk-11.1.2
First thing to *ALWAYS* check is if you have any Asterisk version specific modules (Fax for Asterisk, G.729, etc). Ensure these are not loaded (noload in modules.conf, or simply move them out of the asterisk modules dir). Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Original Message - > After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting > a Segmentation fault. > [root@localhost asterisk-11.1.2]# asterisk -vvc > Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others. > Created by Mark Spencer > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' > for details. > This is free software, with components licensed under the GNU General > Public > License version 2 and other licenses; you are welcome to redistribute > it under > certain conditions. Type 'core show license' for details. > = > == Parsing '/etc/asterisk/asterisk.conf': Found > == Parsing '/etc/asterisk/extconfig.conf': Found > == Parsing '/etc/asterisk/logger.conf': Found > == Manager registered action DBGet > == Manager registered action DBPut > == Manager registered action DBDel > == Manager registered action DBDelTree > == Registered custom function 'MESSAGE' > == Registered custom function 'MESSAGE_DATA' > == Registered application 'MessageSend' > == Manager registered action MessageSend > == Manager registered action DataGet > == Parsing '/etc/asterisk/codecs.conf': Found > Asterisk Dynamic Loader Starting: > == Parsing '/etc/asterisk/modules.conf': Found > == Parsing '/etc/asterisk/dnsmgr.conf': Found > [2013-01-10 14:20:10] ERROR[27062]: config_options.c:512 > aco_process_config: Unable to load config file 'acl.conf' > == Parsing '/etc/asterisk/http.conf': Found > == Manager registered action Ping > == Manager registered action Events > == Manager registered action Logoff > == Manager registered action Login > == Manager registered action Challenge > == Manager registered action Hangup > == Manager registered action Status > == Manager registered action Setvar > == Manager registered action Getvar > == Manager registered action GetConfig > == Manager registered action GetConfigJSON > == Manager registered action UpdateConfig > == Manager registered action CreateConfig > == Manager registered action ListCategories > == Manager registered action Redirect > == Manager registered action Atxfer > == Manager registered action Originate > == Manager registered action Command > == Manager registered action ExtensionState > == Manager registered action PresenceState > == Manager registered action AbsoluteTimeout > == Manager registered action MailboxStatus > == Manager registered action MailboxCount > == Manager registered action ListCommands > == Manager registered action SendText > == Manager registered action UserEvent > == Manager registered action WaitEvent > == Manager registered action CoreSettings > == Manager registered action CoreStatus > == Manager registered action Reload > == Manager registered action CoreShowChannels > == Manager registered action ModuleLoad > == Manager registered action ModuleCheck > == Manager registered action AOCMessage > == Manager registered action Filter > == Registered custom function 'AMI_CLIENT' > == Parsing '/etc/asterisk/manager.conf': Found > == Parsing '/etc/asterisk/manager_humbug.conf': Found > [2013-01-10 14:20:10] NOTICE[27062]: manager.c:7545 __init_manager: > Invalid keyword = in manager.conf > [general] > == Parsing '/etc/asterisk/users.conf': Found > == Parsing '/etc/asterisk/cdr.conf': Found > [2013-01-10 14:20:10] NOTICE[27062]: cdr.c:1613 do_reload: CDR > logging disabled, data will be lost. > -- CEL logging disabled. > == Parsing '/etc/asterisk/udptl.conf': Found > [2013-01-10 14:20:10] WARNING[27062]: udptl.c:1413 > removed_options_handler: t38faxudpec in udptl.conf is no longer > supported; use the t38pt_udptl configuration option in sip.conf > instead. > [2013-01-10 14:20:10] WARNING[27062]: udptl.c:1415 > removed_options_handler: t38faxmaxdatagram in udptl.conf is no > longer supported; value is now supplied by T.38 applications. > Asterisk PBX Core Initializing > Registering builtin applications: > == Registered custom function 'EXCEPTION' > == Registered custom function 'TESTTIME' > [Answer] > == Registered application 'Answer' > [BackGround] > == Registered application 'BackGround' > [Busy] > == Registered application 'Busy' > [Congestion] > == Registered application 'Congestion' > [ExecIfTime] > == Registered application 'ExecIfTime' > [Goto] > == Registered application 'Goto' > [GotoIf] > == Registered application 'GotoIf' > [GotoIfTime] > == Registered application 'GotoIfTime' > [ImportVar] > == Registered application 'ImportVar' > [Hangup] > == Registered application 'Hangup' > [Incomplete] > == Registered application 'Incomplete' > [NoOp] > == Registered application 'NoOp' > [Proceeding] > == Registered applicati
[asterisk-users] Segmentation fault after upgrading from asterisk-10.5.0 to asterisk-11.1.2
After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a Segmentation fault. [root@localhost asterisk-11.1.2]# asterisk -vvc Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found == Parsing '/etc/asterisk/logger.conf': Found == Manager registered action DBGet == Manager registered action DBPut == Manager registered action DBDel == Manager registered action DBDelTree == Registered custom function 'MESSAGE' == Registered custom function 'MESSAGE_DATA' == Registered application 'MessageSend' == Manager registered action MessageSend == Manager registered action DataGet == Parsing '/etc/asterisk/codecs.conf': Found Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found == Parsing '/etc/asterisk/dnsmgr.conf': Found [2013-01-10 14:20:10] ERROR[27062]: config_options.c:512 aco_process_config: Unable to load config file 'acl.conf' == Parsing '/etc/asterisk/http.conf': Found == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Login == Manager registered action Challenge == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action GetConfig == Manager registered action GetConfigJSON == Manager registered action UpdateConfig == Manager registered action CreateConfig == Manager registered action ListCategories == Manager registered action Redirect == Manager registered action Atxfer == Manager registered action Originate == Manager registered action Command == Manager registered action ExtensionState == Manager registered action PresenceState == Manager registered action AbsoluteTimeout == Manager registered action MailboxStatus == Manager registered action MailboxCount == Manager registered action ListCommands == Manager registered action SendText == Manager registered action UserEvent == Manager registered action WaitEvent == Manager registered action CoreSettings == Manager registered action CoreStatus == Manager registered action Reload == Manager registered action CoreShowChannels == Manager registered action ModuleLoad == Manager registered action ModuleCheck == Manager registered action AOCMessage == Manager registered action Filter == Registered custom function 'AMI_CLIENT' == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_humbug.conf': Found [2013-01-10 14:20:10] NOTICE[27062]: manager.c:7545 __init_manager: Invalid keyword = in manager.conf [general] == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/cdr.conf': Found [2013-01-10 14:20:10] NOTICE[27062]: cdr.c:1613 do_reload: CDR logging disabled, data will be lost. -- CEL logging disabled. == Parsing '/etc/asterisk/udptl.conf': Found [2013-01-10 14:20:10] WARNING[27062]: udptl.c:1413 removed_options_handler: t38faxudpec in udptl.conf is no longer supported; use the t38pt_udptl configuration option in sip.conf instead. [2013-01-10 14:20:10] WARNING[27062]: udptl.c:1415 removed_options_handler: t38faxmaxdatagram in udptl.conf is no longer supported; value is now supplied by T.38 applications. Asterisk PBX Core Initializing Registering builtin applications: == Registered custom function 'EXCEPTION' == Registered custom function 'TESTTIME' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [ExecIfTime] == Registered application 'ExecIfTime' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [ImportVar] == Registered application 'ImportVar' [Hangup] == Registered application 'Hangup' [Incomplete] == Registered application 'Incomplete' [NoOp] == Registered application 'NoOp' [Proceeding] == Registered application 'Proceeding' [Progress] == Registered application 'Progress' [RaiseException] == Registered application 'RaiseException' [ResetCDR] == Registered application 'ResetCDR' [Ringing] == Registered application 'Ringing' [SayAlpha] == Registered application 'SayAlpha' [SayDigits] == Registered application 'SayDigits' [SayNumber] == Registered application 'SayNumber' [SayPhonetic] == Regist
Re: [asterisk-users] Segmentation Fault
On 11-08-03 02:01 AM, Richard Zulu wrote: I have used gdb so that I can perform a backtrace however the program executes and exits without a stack thus not helpful. Any help is appreciated! https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://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
[asterisk-users] Segmentation Fault
Hey, I have installed Asterisk 1.8 on slackware 13.1, php, mysql and apache. I am later to install freepbx to help with reporting on VOIP activity. However, after installing asterisk, I am getting a segment fault. My log file shows this: darkstar kernel: asterisk[2660]: segfault at 81c4f ip 77514810 sp 7fffcd48 error 4 in libc-2.11.1.so[77492000+16b000] I have used gdb so that I can perform a backtrace however the program executes and exits without a stack thus not helpful. Any help is appreciated! Richard Zulu Twitter www.twitter.com/richardzulu Skype: zulu.richard * * *There is no place like 127.0.0.1* -- _ -- 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
Re: [asterisk-users] segmentation fault
- "Rilawich Ango" <[EMAIL PROTECTED]> wrote: > Segmentation fault occurs after executing the following cmd. > Dial(SIP/[EMAIL PROTECTED]|35|Ttr) > > Is it a bug and how to fix it? > > Below is the core dump message converted by gdb. > > #0 0x068be1ad in realtime_peer (newpeername=0x1b37844 "10.20.0.1", > sin=0x0) > at chan_sip.c:2547 > #1 0x068becb3 in find_peer (peer=0x1b37844 "10.20.0.1", sin=0x0, > realtime=1) > at chan_sip.c:2676 It is a bug, appears to be the same problem as bug 12362, and the fix is in the pending 1.4.20 release as well as the 1.4 branch in subversion. Jeff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] segmentation fault
Segmentation fault occurs after executing the following cmd. Dial(SIP/[EMAIL PROTECTED]|35|Ttr) Is it a bug and how to fix it? Below is the core dump message converted by gdb. #0 0x068be1ad in realtime_peer (newpeername=0x1b37844 "10.20.0.1", sin=0x0) at chan_sip.c:2547 #1 0x068becb3 in find_peer (peer=0x1b37844 "10.20.0.1", sin=0x0, realtime=1) at chan_sip.c:2676 #2 0x068c0d4b in create_addr (dialog=0x9fb05b8, opeer=0x1b37e24 "10.20.0.1") at chan_sip.c:2902 #3 0x069022d1 in sip_request_call (type=0x1b38010 "SIP", format=256, data=0x1b38b08, cause=0x1b38c08) at chan_sip.c:15992 #4 0x0808c50a in ast_request (type=0x1b38010 "SIP", format=256, data=0x1b38b08, cause=0x1b38c08) at channel.c:2994 #5 0x00df6d91 in dial_exec_full (chan=0x9f91f20, data=0x1b3af38, peerflags=0x1b38e04, continue_exec=0x0) at app_dial.c:1180 #6 0x00dfa3b6 in dial_exec (chan=0x9f91f20, data=0x1b3af38) at app_dial.c:1747 #7 0x080ce1ca in pbx_exec (c=0x9f91f20, app=0x9f37890, data=0x1b3af38) at pbx.c:537 #8 0x080d1f3b in pbx_extension_helper (c=0x9f91f20, con=0x0, context=0x9f92160 "internal-admin", exten=0x9f921b0 "104", priority=4, label=0x0, callerid=0x9f727b8 "200", action=E_SPAWN) at pbx.c:1862 #9 0x080d3280 in ast_spawn_extension (c=0x9f91f20, context=0x9f92160 "internal-admin", exten=0x9f921b0 "104", priority=4, callerid=0x9f727b8 "200") at pbx.c:2317 #10 0x080d37ac in __ast_pbx_run (c=0x9f91f20) at pbx.c:2419 #11 0x080d45c8 in pbx_thread (data=0x9f91f20) at pbx.c:2634 #12 0x08117094 in dummy_start (data=0x9f70fe8) at utils.c:865 #13 0x0076b45b in start_thread () from /lib/libpthread.so.0 #14 0x00bf124e in clone () from /lib/libc.so.6 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Segmentation fault in app_voicemail (ODBC/PSQL problem)
I have been testing with asterisk 1.4.11 and have found a segmentation fault while using voicemail. It happens when I try to forward a voicemail. As soon as I press the option the server crashes. I ran asterisk up inside gdb and got the following stack trace Program received signal SIGSEGV, Segmentation fault. 0x00140adf in CI_read_fields () from /usr/lib/libodbcpsql.so (gdb) bt #0 0x00140adf in CI_read_fields () from /usr/lib/libodbcpsql.so #1 0x001594dc in QR_fetch_tuples () from /usr/lib/libodbcpsql.so #2 0x0014255e in CC_send_query () from /usr/lib/libodbcpsql.so #3 0x0015c87c in SC_execute () from /usr/lib/libodbcpsql.so #4 0x0014bd60 in PG_SQLExecute () from /usr/lib/libodbcpsql.so #5 0x0014bf4d in SQLExecute () from /usr/lib/libodbcpsql.so #6 0x00356cf7 in SQLExecute () from /usr/lib/libodbc.so.1 #7 0x00128e76 in ast_odbc_prepare_and_execute (obj=0x86e76e0, prepare_cb=0x113120 , data=0xb7d83794) at res_odbc.c:96 #8 0x00113511 in realtime_odbc (database=0xb7d838ec "asterisk", table=0xb7d837ec "miniminder.voicemail_users", ap=0xb7d83a04 "C�B") at res_config_odbc.c:148 #9 0x0808c4c7 in ast_load_realtime (family=0x42b12d "voicemail") at config.c:1327 #10 0x0041de88 in find_user (ivm=0xb7d8df60, context=0xb7d8e3cb "miniminder", mailbox=0xb7d8e3c4 "123456") at app_voicemail.c:717 #11 0x0041e5b3 in leave_voicemail (chan=0x878be98, ext=0xb7d9005a "[EMAIL PROTECTED]", options=0xb7d900fc) at app_voicemail.c:2804 #12 0x00425b13 in advanced_options (chan=0x878be98, vmu=0xb7d972c8, vms=0xb7d90208, msg=0, option=1, record_gain=0 '\0') at app_voicemail.c:8111 #13 0x00429ab4 in vm_execmain (chan=0x878be98, data=0xb7d9bf28) at app_voicemail.c:6630 #14 0x080c587e in pbx_extension_helper (c=0x878be98, con=0x0, context=0x878c018 "miniminder", exten=0x878c068 "2000", priority=1, label=0x0, callerid=0x878c358 "123456", action=E_SPAWN) at pbx.c:532 #15 0x080c7566 in __ast_pbx_run (c=0x878be98) at pbx.c:2288 #16 0x080c850e in pbx_thread (data=0x878be98) at pbx.c:2603 #17 0x080f3dbb in dummy_start (data=0x878c8b0) at utils.c:775 #18 0x007d945b in start_thread () from /lib/libpthread.so.0 #19 0x0067c23e in clone () from /lib/libc.so.6 (gdb) step Single stepping until exit from function CI_read_fields, which has no line number information. [Stepped over thread exit] Program terminated with signal SIGSEGV, Segmentation fault. The program no longer exists. Which suggests to me that the problem is in libodbcpgsql, although it might be that asterisk is passing it a bad request. I am using version 2.2.11 of unixODBC and version 8.01.0200 of postgresql-odbc. Can anybody suggest how to take this further? Steve Hindmarch ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segmentation fault
Adam Lovegrove wrote:
> Asterisk is crashing about once a day with segmentation fault.
>
> This is the error..
>
> /usr/sbin/safe_asterisk: line 111: 3482 Segmentation fault (core
> dumped) nice –n $PRIORITY ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS}
>>&/dev/${TTY} Asterisk ended with exit status 139
> Asterisk exited on signal 11.
> Automatically restarting Asterisk.
> mpg123: no process killed
>
> Is this information helpful?
> Can anyone suggest anything?
> Can I provide anymore useful information for troubleshooting?
I think it would be useful if you could describe what the system is
doing when it crashes - if you know. Also, the core dump will probably
help someone diagnose the problem. (but don't send it to the list :-)
/Per Jessen, Zürich
--
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Starting at SFr1/month/user - http://www.spamchek.ch/
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[asterisk-users] Segmentation fault
Asterisk is crashing about once a day with segmentation fault.
This is the error..
/usr/sbin/safe_asterisk: line 111: 3482 Segmentation fault (core
dumped) nice –n $PRIORITY ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS}
&/dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
Is this information helpful?
Can anyone suggest anything?
Can I provide anymore useful information for troubleshooting?
Thanks!
Adam
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[asterisk-users] Segmentation Fault
Hello..I own a server running Slackware 10.2 with kernel 6.1.13 and I tried unsuccessfully to install recently Asterisk 1.4.0. I install all packages but when I execute the command asterisk -vc in order to start asterisk, I get a message Segmentation Fault and the debugging stops suddenly. Does anyone can help me?Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Segmentation fault issue
I to all, I've a segmentation fault while using asterisk relatime conf with mysql db. I've cretate sip_buddies and extensions tables into db and edit res_mysql.conf, extconf.conf without any issues. So when I start asterisk and my phone try to register using sip user configured in my db, asterisk stops with Segmentation fault error. Follow post gdb backtrace 0 0x400337c0 in pthread_setcanceltype () from /lib/libpthread.so.0 #21 0x0805d8de in ast_load_realtime (family=0x666d7464 ) at config.c:994 #22 0x4047cdad in realtime_peer (peername=0xbe7f8891 "101", sin=0x730) at chan_sip.c:1696 #23 0x4046cf67 in find_peer (peer=0xbe7f8891 "101", sin=0x0, realtime=1) at chan_sip.c:1776 #24 0x40485dfd in register_verify (p=0x81944d8, sin=0xbe7fe79c, req=0xbe7fe7ac, uri=0xbe7fe9cd "sip:192.168.1.2", ignore=1718449252) at chan_sip.c:6514 #25 0x404839b7 in handle_request (p=0x81944d8, req=0xbe7fe7ac, sin=0xbe7fe79c, recount=0x666d7464, nounlock=0x666d7464) at chan_sip.c:11083 #26 0x4048150d in sipsock_read (id=0x813ed80, fd=15, events=1, ignore=0x0) at chan_sip.c:11377 #27 0x080558dd in ast_io_wait (ioc=0x8162320, howlong=1718449252) at io.c:284 #28 0x404776a9 in do_monitor (data=0x0) at chan_sip.c:11536 #29 0x40034cc4 in pthread_detach () from /lib/libpthread.so.0 #30 0x40201037 in clone () from /lib/libc.so.6 any suggestion about? Thanks 4 all, -- * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Segmentation fault asterisk realtime problem
Hi to all, I've a segmentation fault while using asterisk relatime conf with mysql db. I've cretate sip_buddies and extensions tables into db and edit res_mysql.conf, extconf.conf without any issues. So when I start asterisk and my phone try to register using sip user configured in my db, asterisk stops with Segmentation fault error. Follow post gdb backtrace 0 0x400337c0 in pthread_setcanceltype () from /lib/libpthread.so.0 #21 0x0805d8de in ast_load_realtime (family=0x666d7464 ) at config.c:994 #22 0x4047cdad in realtime_peer (peername=0xbe7f8891 "101", sin=0x730) at chan_sip.c:1696 #23 0x4046cf67 in find_peer (peer=0xbe7f8891 "101", sin=0x0, realtime=1) at chan_sip.c:1776 #24 0x40485dfd in register_verify (p=0x81944d8, sin=0xbe7fe79c, req=0xbe7fe7ac, uri=0xbe7fe9cd "sip:192.168.1.2", ignore=1718449252) at chan_sip.c:6514 #25 0x404839b7 in handle_request (p=0x81944d8, req=0xbe7fe7ac, sin=0xbe7fe79c, recount=0x666d7464, nounlock=0x666d7464) at chan_sip.c:11083 #26 0x4048150d in sipsock_read (id=0x813ed80, fd=15, events=1, ignore=0x0) at chan_sip.c:11377 #27 0x080558dd in ast_io_wait (ioc=0x8162320, howlong=1718449252) at io.c:284 #28 0x404776a9 in do_monitor (data=0x0) at chan_sip.c:11536 #29 0x40034cc4 in pthread_detach () from /lib/libpthread.so.0 #30 0x40201037 in clone () from /lib/libc.so.6 any suggestion about? Thanks 4 all, -- * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segmentation fault on Asteriskstartup:res_config_mysql.so problem?
Did you do a make && make install for add-ons BEFORE doing
so for asterisk?
If so try asterisk first and when all is installed install
add-ons.
--
I tried a make clean && make && make install for asterisk and then for
asterisk-addons but am still getting the segmentation fault on asterisk
startup. rm res_config_mysql.so allows Asterisk to start.
Still trying...
mkdir /usr/lib/asterisk.backup.20060928
mv /usr/lib/asterisk/* /usr/lib/asterisk.backup.20060928
mkdir /usr/include/asterisk.backup.20060928
mv /usr/include/asterisk/* /usr/include/asterisk.backup.20060928/
cd /usr/src/asterisk-1.2.12.1
make clean && make && make install
cd /usr/src/asterisk-addons-1.2.4
perl -p -i.bak -e
's/CFLAGS.*D_GNU_SOURCE/CFLAGS+=-D_GNU_SOURCE\nCFLAGS+=-DMYSQL_LOGUNIQUEID/'
Makefile
make clean && make && make install
Install logs look fine.
STARTING ASTERISK
/usr/sbin/safe_asterisk: line 40: 6631 Segmentation fault (core
dumped) asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 40: 6690 Segmentation fault (core
dumped) asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
rm res_config_mysql.so allows Asterisk to start.
Any advice appreciated.
Cameron
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Re: [asterisk-users] Segmentation fault on Asterisk startup:res_config_mysql.so problem?
Did you do a make && make install for add-ons BEFORE doing so for asterisk? If so try asterisk first and when all is installed install add-ons. -- I tried a make clean && make && make install for asterisk and then for asterisk-addons but am still getting the segmentation fault on asterisk startup. rm res_config_mysql.so allows Asterisk to start. Any other suggestions appreciated. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segmentation fault on Asterisk startup: res_config_mysql.so problem?
On 16:04, Sun 24 Sep 06, kjcsb wrote:
> When Asterisk starts I get a Segmentation fault
> /usr/sbin/safe_asterisk: line 40: 30548 Segmentation fault (core
> dumped) asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} Asterisk ended with exit status 139
> Asterisk exited on signal 11.
>
> If I remove /usr/lib/asterisk/modules/res_config_mysql.so Asterisk starts
> normally.
>
> tail /var/log/asterisk/full.log
> Sep 24 15:46:05 VERBOSE[30608] logger.c: == Parsing
> '/etc/asterisk/res_mysql.conf': Sep 24 15:46:05 VERBOSE[30608] logger.c:
> == Parsing '/etc/asterisk/res_mysql.conf': Found
> Sep 24 15:46:05 WARNING[30608] res_config_mysql.c: MySQL RealTime: No
> database socket found, using '/tmp/mysql.sock' as default.
> Sep 24 15:46:05 DEBUG[30608] res_config_mysql.c: MySQL RealTime Host:
> 127.0.0.1
> Sep 24 15:46:05 DEBUG[30608] res_config_mysql.c: MySQL RealTime Port: 3306
> Sep 24 15:46:05 DEBUG[30608] res_config_mysql.c: MySQL RealTime User: root
> Sep 24 15:46:05 DEBUG[30608] res_config_mysql.c: MySQL RealTime Password:
> password
>
> vi /etc/asterisk/res_mysql.conf
> [general]
> dbhost = 127.0.0.1
> dbname = asterisk
> dbuser = root
> dbpass = password
> dbport = 3306
> ;dbsock = /var/lib/mysql/mysql.sock
>
> If I uncomment the dbsock line I get the same result (although the database
> socket warning is not displayed in the log file).
>
> I am using MySQL for CDR logging so I don't think it's a MySQL problem.
>
> Asterisk 1.2.12.1
> Asterisk addon 1.2.4
>
> When I install Asterisk I receive a warning:
> Your Asterisk modules directory, located at /usr/lib/asterisk/modules
> contains modules that were not installed by this version of Asterisk.
>
> However I cleared out the /usr/lib/asterisk/modules directory before make
> clean && make && make install for both add-ons and asterisk so I'm a bit
> mystified by that.
>
Did you do a make && make install for add-ons BEFORE doing
so for asterisk?
If so try asterisk first and when all is installed install
add-ons.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD
"Why is it drug addicts and computer afficionados are both called users?"
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[asterisk-users] Segmentation fault on Asterisk startup: res_config_mysql.so problem?
When Asterisk starts I get a Segmentation fault
/usr/sbin/safe_asterisk: line 40: 30548 Segmentation fault (core
dumped) asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
If I remove /usr/lib/asterisk/modules/res_config_mysql.so Asterisk starts
normally.
tail /var/log/asterisk/full.log
Sep 24 15:46:05 VERBOSE[30608] logger.c: == Parsing
'/etc/asterisk/res_mysql.conf': Sep 24 15:46:05 VERBOSE[30608] logger.c:
== Parsing '/etc/asterisk/res_mysql.conf': Found
Sep 24 15:46:05 WARNING[30608] res_config_mysql.c: MySQL RealTime: No
database socket found, using '/tmp/mysql.sock' as default.
Sep 24 15:46:05 DEBUG[30608] res_config_mysql.c: MySQL RealTime Host:
127.0.0.1
Sep 24 15:46:05 DEBUG[30608] res_config_mysql.c: MySQL RealTime Port: 3306
Sep 24 15:46:05 DEBUG[30608] res_config_mysql.c: MySQL RealTime User: root
Sep 24 15:46:05 DEBUG[30608] res_config_mysql.c: MySQL RealTime Password:
password
vi /etc/asterisk/res_mysql.conf
[general]
dbhost = 127.0.0.1
dbname = asterisk
dbuser = root
dbpass = password
dbport = 3306
;dbsock = /var/lib/mysql/mysql.sock
If I uncomment the dbsock line I get the same result (although the database
socket warning is not displayed in the log file).
I am using MySQL for CDR logging so I don't think it's a MySQL problem.
Asterisk 1.2.12.1
Asterisk addon 1.2.4
When I install Asterisk I receive a warning:
Your Asterisk modules directory, located at /usr/lib/asterisk/modules
contains modules that were not installed by this version of Asterisk.
However I cleared out the /usr/lib/asterisk/modules directory before make
clean && make && make install for both add-ons and asterisk so I'm a bit
mystified by that.
Could anyone suggest further checks I could do?
Thanks
Cameron
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Re: [Asterisk-Users] segmentation fault
I had the same problem when I did an SVN of the latest version of the
1.2 branch this past Friday (2/10) and re-made zaptel, libpri and
asterisk. I don't if someone goofed when updating the supposed stable
Asterisk 1.2 branch or what. But here's what I did to fix it...
I'm using Asterisk at Home. To get back to a working system I first
made a backup of everything through the AMP interface. I made backups
of my zaptel.conf and zapata.conf files to a separate location. Then, I
downloaded the tar version of AAH 2.5 (instead of the ISO). I followed
the instructions to untar and install AAH. Because AAH doesn't really
have a 100% working restore feature, what I had to do was:
1) go ahead and use the restore function in the AMP interface -
restoring the backup I had just made.
2) Go through every single configuration window within the AMP
interface and click 'submit' to re-apply the settings one by one. This
really only takes like 10 minutes.
3) Then I clicked the red bar along the top to apply the change in
settings.
4) In the process of installing itself, AAH will blow away your
zaptel.conf and zapata.conf files that you spent so long just getting
right. That's why you made backup copies of those to another directory.
;) Restore those files manually.
5) In order to get my T1 actually functioning I had to not only
shutdown the server but pull the plug on it for a few minutes and then
plug it back in/hit the on button.
Not sure if you're using AAH as well - but hope this helps someone else
who ended up in the same position as I did late Friday night.
Best of Luck,
Sascha
On Mon, 13 Feb 2006 10:02:28 -0500, Patrick Fortin wrote:
Hi
Asterisk died this morning with this message
safe_asterisk: line 83: 6828 Segmentation fault (core dumped)
asterisk ${CLIARGS} ${ASTARGS} 1>&/dev/${TTY}
Re: [Asterisk-Users] segmentation fault
On Mon, 13 Feb 2006 10:02:28 -0500, Patrick Fortin <[EMAIL PROTECTED]>
wrote:
Hi
Asterisk died this morning with this message
safe_asterisk: line 83: 6828 Segmentation fault (core dumped)
asterisk ${CLIARGS} ${ASTARGS} 1>&/dev/${TTY}
Any idea what is the problem ?
You should post this on the bugtracker (bugs.digium.com) along with a back
trace. See asterisk-sources/doc/README.backtrace for info on how to do a
backtrace.
Justin Tunney
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[Asterisk-Users] segmentation fault
Hi
Asterisk died this morning with this message
safe_asterisk: line 83: 6828 Segmentation fault (core dumped)
asterisk ${CLIARGS} ${ASTARGS} 1>&/dev/${TTY}
Any idea what is the problem ?
here is a show channels before the crash
SIP/131-f5ad (None) Ringing AppDial((Outgoing
Line))
SIP/123-8bc1 [EMAIL PROTECTED]:1 RingDial(SIP/131|16|tr)
Zap/11-1 [EMAIL PROTECTED]:2Up Queue(reception|t|||300)
Zap/10-1 [EMAIL PROTECTED]:2Up Queue(reception|t|||300)
Zap/3-1 [EMAIL PROTECTED]:1 Up Bridged
Call(SIP/137-adb6)
SIP/137-adb6 [EMAIL PROTECTED]:2
Up Dial(Zap/G1/915143334233)
Zap/8-1 [EMAIL PROTECTED]:2 Up Queue(support|t|||300)
Zap/7-1 [EMAIL PROTECTED]:2Up Queue(reception|t|||300)
SIP/141-178a [EMAIL PROTECTED]:1 Up
AgentCallbackLogin(141|@zapout
SIP/141-f5f6 [EMAIL PROTECTED]:1 Up
AgentCallbackLogin(141|@zapout
SIP/141-5371 [EMAIL PROTECTED]:1 Up
AgentCallbackLogin(141|@zapout
Zap/4-1 [EMAIL PROTECTED]:2Up Queue(reception|t|||300)
Agent/111[EMAIL PROTECTED]:1 Down(None)
Local/[EMAIL PROTECTED]
[EMAIL PROTECTED]:1Ring(None)
Local/[EMAIL PROTECTED]
[EMAIL PROTECTED]:1 Down(None)
Zap/6-1 [EMAIL PROTECTED]:2Up Queue(reception|t|||300)
Zap/2-1 [EMAIL PROTECTED]:2Up Queue(reception|t|||300)
The queue application seemed to be crashed because I got no output from a
show agents or a show queues
Patrick
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[Asterisk-Users] Segmentation fault
Hiya - I have compiled: zaptel-1.0.9.1 libpri-1.0.9 asterisk 1.0.9 asterisk-addons-1.0.9 all new, all make'd and installed... When I run asterisk, I get the following: Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf Segmentation fault Any ideas what I have done wrong? thank you! Álainn * "The cheese-mites asked how the cheese got there, And warmly debated the matter; The orthodox said that it came from the air, And the heretics said from the platter." Anon. ___ 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] Segmentation fault {Writer given gnu-lashing}
On Sun, 2005-02-20 at 22:59 +0100, Julius Schwartzenberg wrote: > I'm using a pretty old system and I have good experiences with Slackware > on other systems. Here are the specs of the system I'm using: > IBM/Cyrix PR-200 (@150MHz), 64MB RAM, two HDs which are combined ~2GB. Here is your trouble. The Cyrix chip is what is the newer Via chipsets are based on. It isn't a real pentium chipset and needs to get tuned down via the CFLAGS to 586 or lower. You will probably hit the limits of that machine really quickly. You may want to find a slightly better machine for testing. -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] Segmentation fault {Writer given gnu-lashing}
Thanks a lot for your message. Race Vanderdecken schreef: Ouch, Do you know how to use gdb, the Gnu Debugger? That will give you a clue as to where the segmentation fault is coming from. No, I once used it being instructed exactly by a developer to solve a problem in Dosemu, but I never did anything else with it. I understand that I need to recompile Asterisk with debugging support. Could you give me some pointers on what to do next? Good, then let me move on to the insults and ranting. 1. Why are you running on Slackware? Are you trying to prove a point or just enjoy being frustrated? Open Source is like "Broad Spectrum Pesticide", it works but your results may vary and you may end up killing your lawn. I'm using a pretty old system and I have good experiences with Slackware on other systems. Here are the specs of the system I'm using: IBM/Cyrix PR-200 (@150MHz), 64MB RAM, two HDs which are combined ~2GB. 2. The dearth of information of value in your posting is amazing. I went to http://www.automated.it/guidetoasterisk.htm (a good start, good effort Mr. Powell.) As stated above, you life might be easier using FEDORA, not an endorsement of Red Hat, rather a plea for a unified Linux base (please don't say Debian, self-installing the micro-chip in my head was easier.) (it is the new Anti-AMD Intel Rantino chip for those interested.) I've never used Fedora on older systems, but I thought it wouldn't run very well on the system I'm using. (Good thing you don't have an Anti-IBM chip ;) 3. "I've never installed or used Asterisk before, so I do not know much about it." 1. What is your goal with installing Asterisk? We have about 8 telephones that use the plain telephone system to call each other and externally. Some of them are analog and others are digital (ISDN). I've also still got the old ISDN card from before we had ADSL. (Eicon Diva 2.01 ISA, seems to work with the hisax module.) Since I read that Asterisk worked with any ISDN adapter that was supported by ISDN4Linux, I thought it might be possible to hook it up in such a way that the phones could call the Asterisk system and that Asterisk would forward the call to a computer (and maybe even over the internet). Also the other way around would be neat. 2. Do you have Digium or other hardware installed? No. Only the ISDN adapter. 3. Are you running SIP/H323/MGCP? No. I've experimented with SIP before, but only with a softphone, using an account from SIPPhone.com. It would be nice if I could call my Asterisk system using SIP! 4. Did you modify any files? None from Asterisk. 4. What was the last thing on the *CLI> screen before the seg fault? The command to run Asterisk. It immediatly gives the error when I try to run it. Is Asterisk able to do, what I thought it would do or am I just messing? Come on Mr. Caesar throw us a bone here. All Hail, Race "The Tyrant" Vanderdecken Thanks again, Julius ___ 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] Segmentation fault {Writer given gnu-lashing}
[EMAIL PROTECTED] wrote:
> I have no problem with Slackware,
Me neither. I learned Linux with Slack. Found it to be extremely
friendly. And that was 10 years ago. Last time I chacked, it was still
friendly (and not at all GUI, unless you want it served that way)
> But when you are learning to drive a car you should first try
> a Chevy with an automatic transmission first before strapping
> on a 6 speed Ferrari.
Popular opinion holds that people who learn to drive standard first
generally end up being better drivers. And why wouldn't you want to
learn on a Ferrari since you can get one for free!?!
> Humor helps in teaching and getting a person to step out of a
> rut they are having a problem in and gives them a chance to
> rethink what might be going on.
Ya, but humour should be dispensed carefully, lest offence be given.
> Remember, my goal is to reduce the number of variables in the system.
The problem I see with Fedora is that you can install it successfully
without learning anything about Linux. Slackware is rather good for
learning Linux, because it is friendly and helpful, but still expects
you to make the decisions. I'd argue that a familiarity with the shell
is going to be essential for even a basic Asterisk install. It's not a
pre-qualifier so much as an essential skill.
LOL! You're just bored and are trolling for a holy war, eh? Well, I
guess we gotta shake off these Febraury blah's somehow.
GENTOO IS FOR WANNABE NEWBIES!!! (that oughta stir things up)
--
Jim Van Meggelen
[EMAIL PROTECTED]
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Andrew Kohlsmith
> Sent: Sunday, February 20, 2005 1:39 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [Asterisk-Users] Segmentation fault {Writer
> given gnu-lashing}
>
> On February 20, 2005 01:11 pm, Race Vanderdecken wrote:
>> 1. Why are you running on Slackware?
>> Are you trying to prove a point or just enjoy being frustrated?
>> Open Source is like "Broad Spectrum Pesticide", it works but your
>> results may vary and you may end up killing your lawn.
>
> Got a problem with Slackware? It works *very* well with Asterisk.
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005
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Re: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}
On Sun, 2005-02-20 at 13:51 -0500, Paul wrote: > Or maybe a double fool because he also disrespected Debian GNU/Linux in > his reply. *And* recommended Fedora, which makes it triple. I just dumped FC3 and replaced it with Debian because Fedora's kernels constantly gave me issues, e.g. with proprietary AVM kernel drivers which didn't even work. On the other hand, no probs whatsoever with Debian. ___ 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] Segmentation fault {Writer given gnu-lashing}
I have no problem with Slackware,
But when you are learning to drive a car you should first try a Chevy
with an automatic transmission first before strapping on a 6 speed
Ferrari.
Humor helps in teaching and getting a person to step out of a rut they
are having a problem in and gives them a chance to rethink what might be
going on.
Remember, my goal is to reduce the number of variables in the system.
Race
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Sunday, February 20, 2005 1:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [Asterisk-Users] Segmentation fault {Writer given
gnu-lashing}
On February 20, 2005 01:11 pm, Race Vanderdecken wrote:
> 1. Why are you running on Slackware?
> Are you trying to prove a point or just enjoy being frustrated?
> Open Source is like "Broad Spectrum Pesticide", it works but
> your results may vary and you may end up killing your lawn.
Got a problem with Slackware? It works *very* well with Asterisk.
-A.
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Re: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}
Brian Capouch wrote: Race Vanderdecken wrote: Good, then let me move on to the insults and ranting. 1. Why are you running on Slackware? Are you trying to prove a point or just enjoy being frustrated? Open Source is like "Broad Spectrum Pesticide", it works but your results may vary and you may end up killing your lawn. Why do you not follow Ann Landers simple adage, "Better to keep one's mouth shut and be thought a fool, than to open it and remove any doubt?" Or maybe a double fool because he also disrespected Debian GNU/Linux in his reply. Is ignorance really bliss? ___ 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] Segmentation fault {Writer given gnu-lashing}
Because I am more civilized?
By the way, it was Samuel Clemens's "fool..." quote, who stole it from
Mr. Lincoln, who stole if from Confucius (another educational Tyrant.)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Capouch
Sent: Sunday, February 20, 2005 1:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Segmentation fault {Writer given
gnu-lashing}
Race Vanderdecken wrote:
> Good, then let me move on to the insults and ranting.
>
> 1. Why are you running on Slackware?
> Are you trying to prove a point or just enjoy being frustrated?
> Open Source is like "Broad Spectrum Pesticide", it works but
> your results may vary and you may end up killing your lawn.
>
Why do you not follow Ann Landers simple adage, "Better to keep one's
mouth shut and be thought a fool, than to open it and remove any doubt?"
B.
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Re: [Asterisk-Users] Segmentation fault {Writer given gnu-lashing}
On February 20, 2005 01:11 pm, Race Vanderdecken wrote: > 1. Why are you running on Slackware? > Are you trying to prove a point or just enjoy being frustrated? > Open Source is like "Broad Spectrum Pesticide", it works but > your results may vary and you may end up killing your lawn. Got a problem with Slackware? It works *very* well with Asterisk. -A. ___ 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] Segmentation fault {Writer given gnu-lashing}
Race Vanderdecken wrote: Good, then let me move on to the insults and ranting. 1. Why are you running on Slackware? Are you trying to prove a point or just enjoy being frustrated? Open Source is like "Broad Spectrum Pesticide", it works but your results may vary and you may end up killing your lawn. Why do you not follow Ann Landers simple adage, "Better to keep one's mouth shut and be thought a fool, than to open it and remove any doubt?" B. ___ 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] Segmentation fault {Writer given gnu-lashing}
Ouch, Do you know how to use gdb, the Gnu Debugger? That will give you a clue as to where the segmentation fault is coming from. Good, then let me move on to the insults and ranting. 1. Why are you running on Slackware? Are you trying to prove a point or just enjoy being frustrated? Open Source is like "Broad Spectrum Pesticide", it works but your results may vary and you may end up killing your lawn. 2. The dearth of information of value in your posting is amazing. I went to http://www.automated.it/guidetoasterisk.htm (a good start, good effort Mr. Powell.) As stated above, you life might be easier using FEDORA, not an endorsement of Red Hat, rather a plea for a unified Linux base (please don't say Debian, self-installing the micro-chip in my head was easier.) (it is the new Anti-AMD Intel Rantino chip for those interested.) 3. "I've never installed or used Asterisk before, so I do not know much about it." 1. What is your goal with installing Asterisk? 2. Do you have Digium or other hardware installed? 3. Are you running SIP/H323/MGCP? 4. Did you modify any files? 4. What was the last thing on the *CLI> screen before the seg fault? Come on Mr. Caesar throw us a bone here. All Hail, Race "The Tyrant" Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Schwartzenberg Sent: Sunday, February 20, 2005 11:47 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Segmentation fault Hi, I'm trying to set up a fresh system for use with Asterisk. I've never installed or used Asterisk before, so I do not know much about it. I'm using Slackware Linux 10.1 and followed this guide: http://www.automated.it/guidetoasterisk.htm When I try to run asterisk though, at the point the guide suggests to try it, I get 'Segmentation fault'. Any idea what to do? Are there any known problems with Slackware and Asterisk? Thanks in advance, Julius ___ 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
[Asterisk-Users] Segmentation fault
Hi, I'm trying to set up a fresh system for use with Asterisk. I've never installed or used Asterisk before, so I do not know much about it. I'm using Slackware Linux 10.1 and followed this guide: http://www.automated.it/guidetoasterisk.htm When I try to run asterisk though, at the point the guide suggests to try it, I get 'Segmentation fault'. Any idea what to do? Are there any known problems with Slackware and Asterisk? Thanks in advance, Julius ___ 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] Segmentation Fault after Digitnetwork X100Pinstall
- Original Message - From: "Steven Critchfield" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, January 22, 2005 4:54 AM Subject: Re: [Asterisk-Users] Segmentation Fault after Digitnetwork X100Pinstall On Sat, 2005-01-22 at 04:38 -0600, Rich Adamson wrote: > > > Shouldn't you contact your vendor for support and not a different > > > vendors support channel? > > > > > As far I know, although Digium hosts the asterisk-users list and > > supports the Asterisk development, Asterisk is still a GPL open > > source > > project and asterisk-list is not a Digium support channel. Asking > > questions about a vendor other than Digium is no difference than > > asking questions about Digium hardware. > > Asking about a knock off vendor is grating on many peoples nerves. > Asking about a problem caused by a knock of vendor is VERY grating. Why > should those who are directly supporting Digium by helping to run this > list as well as Digium be charged with having to do support for other > companies? I'm not the OP and do agree with supporting digium where possible, but it would appear that digium is no longer selling the x100p card (or at least its removed from the web site). Given some folks only need/want/aford a single fxo port, is this actually considered a knock-off at this time? Isn't a TDM01b a single port FXO? It also has the benefit of being able to scale up a little bit before having slot/interupt problems. This really is an edge case. If the original vendor had to do the support, they couldn't sell the parts at such a low price point. If the price rose to the amount to offer proper support, there wouldn't be much distinction between them and Digium. Digium would then likely get the sale as it is a feel good action. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Just for FYI Digit Networks offers 30 days of support for the clone X100P. Matter of fact their support is great. I purchased 4 cards from them to set up testing in my lab. I wouldn't use the clones in production use, but for some home users, testing and such they are inexpensive and well just inexpensive. Why spend $133 on a card for a low end machine when you can spend $30. Not everyone has a budget for tinkering around and home use. ___ 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] Segmentation Fault after Digitnetwork X100P install
On Sat, 2005-01-22 at 04:38 -0600, Rich Adamson wrote: > > > > Shouldn't you contact your vendor for support and not a different > > > > vendors support channel? > > > > > > > As far I know, although Digium hosts the asterisk-users list and > > > supports the Asterisk development, Asterisk is still a GPL open source > > > project and asterisk-list is not a Digium support channel. Asking > > > questions about a vendor other than Digium is no difference than > > > asking questions about Digium hardware. > > > > Asking about a knock off vendor is grating on many peoples nerves. > > Asking about a problem caused by a knock of vendor is VERY grating. Why > > should those who are directly supporting Digium by helping to run this > > list as well as Digium be charged with having to do support for other > > companies? > > I'm not the OP and do agree with supporting digium where possible, > but it would appear that digium is no longer selling the x100p card > (or at least its removed from the web site). Given some folks only > need/want/aford a single fxo port, is this actually considered a > knock-off at this time? Isn't a TDM01b a single port FXO? It also has the benefit of being able to scale up a little bit before having slot/interupt problems. This really is an edge case. If the original vendor had to do the support, they couldn't sell the parts at such a low price point. If the price rose to the amount to offer proper support, there wouldn't be much distinction between them and Digium. Digium would then likely get the sale as it is a feel good action. -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] Segmentation Fault after Digitnetwork X100P install
> > > Shouldn't you contact your vendor for support and not a different > > > vendors support channel? > > > > > As far I know, although Digium hosts the asterisk-users list and > > supports the Asterisk development, Asterisk is still a GPL open source > > project and asterisk-list is not a Digium support channel. Asking > > questions about a vendor other than Digium is no difference than > > asking questions about Digium hardware. > > Asking about a knock off vendor is grating on many peoples nerves. > Asking about a problem caused by a knock of vendor is VERY grating. Why > should those who are directly supporting Digium by helping to run this > list as well as Digium be charged with having to do support for other > companies? I'm not the OP and do agree with supporting digium where possible, but it would appear that digium is no longer selling the x100p card (or at least its removed from the web site). Given some folks only need/want/aford a single fxo port, is this actually considered a knock-off at this time? ___ 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] Segmentation Fault after Digitnetwork X100P install
On Fri, 2005-01-21 at 09:55 -0800, Asterisk List wrote: > On Thu, 20 Jan 2005 22:41:08 -0600, Steven Critchfield > <[EMAIL PROTECTED]> wrote: > > > > Shouldn't you contact your vendor for support and not a different > > vendors support channel? > > > As far I know, although Digium hosts the asterisk-users list and > supports the Asterisk development, Asterisk is still a GPL open source > project and asterisk-list is not a Digium support channel. Asking > questions about a vendor other than Digium is no difference than > asking questions about Digium hardware. Asking about a knock off vendor is grating on many peoples nerves. Asking about a problem caused by a knock of vendor is VERY grating. Why should those who are directly supporting Digium by helping to run this list as well as Digium be charged with having to do support for other companies? arghhh. -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] Segmentation Fault after Digitnetwork X100P install
On Thu, 20 Jan 2005 22:41:08 -0600, Steven Critchfield <[EMAIL PROTECTED]> wrote: > > Shouldn't you contact your vendor for support and not a different > vendors support channel? > As far I know, although Digium hosts the asterisk-users list and supports the Asterisk development, Asterisk is still a GPL open source project and asterisk-list is not a Digium support channel. Asking questions about a vendor other than Digium is no difference than asking questions about Digium hardware. --JJL44 ___ 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] Segmentation Fault after Digitnetwork X100Pinstall
Shouldn't you contact your vendor for support and not a different vendors support channel? Um, I didn't think this was a Digium support channel. I thought this was an Asterisk Users channel. Seems to me the question should be fair game. (Sorry I don't have an answer to your question, though, Dave). regards, Paul - Original Message - From: "Steven Critchfield" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, January 20, 2005 9:41 PM Subject: Re: [Asterisk-Users] Segmentation Fault after Digitnetwork X100Pinstall On Fri, 2005-01-21 at 16:57 +1300, Dave Green wrote: I've just installed a Digitnetworks X100P clone in my * server and run the install script for the voicepet-single-x100p tarball. The install appeared to run OK with modprobe wcfxo successful and the ztcfg reporting "Channel 01: FXS Kewlstart (Default) (Slaves: 01)". When I try to start * though I get a segmentation fault after loading res_features.so. I discovered that the Digitnetwork install script seems to modify all of the .conf files, leaving .conf.old copies. I tried moving the .conf.old files back to .conf but am still get the seg fault. Shouldn't you contact your vendor for support and not a different vendors support channel? Another company with retarded disclaimers sending to a KNOWN publicly archived mailing list. Fix the problem or be ridiculed regularly for the stupidity. CAUTION: This message and any attachments contain privileged and confidential information. If you are not the intended recipient of this message, you are hereby notified that any use, dissemination, distribution or reproduction of this message is prohibited. If you have received this message in error please notify the sender immediately via email and then destroy this message and any attachments. Any views expressed in this message are those of the individual sender and may not necessarily reflect the views of Winstone Pulp International Ltd. -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] Segmentation Fault after Digitnetwork X100P install
On Fri, 2005-01-21 at 16:57 +1300, Dave Green wrote: > I've just installed a Digitnetworks X100P clone in my * server and run > the install script for the voicepet-single-x100p tarball. The install > appeared to run OK with modprobe wcfxo successful and the ztcfg > reporting "Channel 01: FXS Kewlstart (Default) (Slaves: 01)". > > When I try to start * though I get a segmentation fault after loading > res_features.so. I discovered that the Digitnetwork install script seems > to modify all of the .conf files, leaving .conf.old copies. I tried > moving the .conf.old files back to .conf but am still get the seg fault. Shouldn't you contact your vendor for support and not a different vendors support channel? Another company with retarded disclaimers sending to a KNOWN publicly archived mailing list. Fix the problem or be ridiculed regularly for the stupidity. > > CAUTION: > This message and any attachments contain privileged and confidential > information. If you are not the intended recipient of this message, you > are hereby notified that any use, dissemination, distribution or > reproduction of this message is prohibited. If you have received this > message in error please notify the sender immediately via email and then > destroy this message and any attachments. > > Any views expressed in this message are those of the individual sender > and may not necessarily reflect the views of Winstone Pulp International > Ltd. -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] Segmentation Fault after Digitnetwork X100P install
Henry Devito wrote: Just download 1.0.3, don't use the voicepet scripts. they are versions 0.7 and 0.9 of *. If it is a true clone from digit networks it will work fine with 1.0.3. I just set up 3 different machines with the cards from digit networks. Henry Yes, thanks for that. After posting I wondered if the script installed the whole * package, so I recompiled/installed the cvs I'd downloaded and after a bit more fiddling have managed ot get it working. Dave CAUTION: This message and any attachments contain privileged and confidential information. If you are not the intended recipient of this message, you are hereby notified that any use, dissemination, distribution or reproduction of this message is prohibited. If you have received this message in error please notify the sender immediately via email and then destroy this message and any attachments. Any views expressed in this message are those of the individual sender and may not necessarily reflect the views of Winstone Pulp International Ltd. ___ 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] Segmentation Fault after Digitnetwork X100P install
Just download 1.0.3, don't use the voicepet scripts. they are versions 0.7 and 0.9 of *. If it is a true clone from digit networks it will work fine with 1.0.3. I just set up 3 different machines with the cards from digit networks. Henry - Original Message - From: "Dave Green" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, January 20, 2005 9:57 PM Subject: [Asterisk-Users] Segmentation Fault after Digitnetwork X100P install I've just installed a Digitnetworks X100P clone in my * server and run the install script for the voicepet-single-x100p tarball. The install appeared to run OK with modprobe wcfxo successful and the ztcfg reporting "Channel 01: FXS Kewlstart (Default) (Slaves: 01)". When I try to start * though I get a segmentation fault after loading res_features.so. I discovered that the Digitnetwork install script seems to modify all of the .conf files, leaving .conf.old copies. I tried moving the .conf.old files back to .conf but am still get the seg fault. Has anyone else run into this kind of problem ? The * server has been running fine up until now. I can't get it to show the * version right now, but compiled from CVS around 11 Jan. Dave CAUTION: This message and any attachments contain privileged and confidential information. If you are not the intended recipient of this message, you are hereby notified that any use, dissemination, distribution or reproduction of this message is prohibited. If you have received this message in error please notify the sender immediately via email and then destroy this message and any attachments. Any views expressed in this message are those of the individual sender and may not necessarily reflect the views of Winstone Pulp International Ltd. ___ 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
[Asterisk-Users] Segmentation Fault after Digitnetwork X100P install
I've just installed a Digitnetworks X100P clone in my * server and run the install script for the voicepet-single-x100p tarball. The install appeared to run OK with modprobe wcfxo successful and the ztcfg reporting "Channel 01: FXS Kewlstart (Default) (Slaves: 01)". When I try to start * though I get a segmentation fault after loading res_features.so. I discovered that the Digitnetwork install script seems to modify all of the .conf files, leaving .conf.old copies. I tried moving the .conf.old files back to .conf but am still get the seg fault. Has anyone else run into this kind of problem ? The * server has been running fine up until now. I can't get it to show the * version right now, but compiled from CVS around 11 Jan. Dave CAUTION: This message and any attachments contain privileged and confidential information. If you are not the intended recipient of this message, you are hereby notified that any use, dissemination, distribution or reproduction of this message is prohibited. If you have received this message in error please notify the sender immediately via email and then destroy this message and any attachments. Any views expressed in this message are those of the individual sender and may not necessarily reflect the views of Winstone Pulp International Ltd. ___ 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] Segmentation fault
Hi,
I
have just got to the office and now * is giving me the following error:
-- Executing Macro("SIP/204-df20",
"dial|CAPI/111222:b130") in new stack
-- Executing DBput("SIP/204-df20",
"RepeatDial/204=130") in new stack
-- DBput: family=RepeatDial, key=204, value=130
-- Executing NoOp("SIP/204-df20",
"") in new stack
-- Executing Dial("SIP/204-df20",
"CAPI/111222:b130|30") in new stack
Ouch ... error while writing audio data: : Broken pipe
Segmentation fault
I
have tried everything, putting an old conf downloading the new head, but
nothing. Friday night everything was working fine. I have even made a
Shutdown/power-up procedure but still no success.
My ISDN card is a Fritz PCI Card. Any help is surely
appreciated since I am completely lost on this one.
;
; CAPI config
;
;
[general]
nationalprefix=11
internationalprefix=55
rxgain=0.0
txgain=0.0
[interfaces]
msn=111222
incomingmsn=*
controller=1
softdtmf=1
accountcode=capi
context=from-capi
echosquelch=1
;echocancel=yes
;echotail=32
callgroup=1
;deflect=12345678
devices=2
isdnmode=ptp
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[Asterisk-Users] Segmentation Fault (core dumped)
i am facing unusual and wiered error in asterisk using Realtime MYSQL driver . Asterisk runs well and smoothly with absoulutely no error or warning but everytime i power-on my sip-phone ,booting, initializes and then asterisk suddenly quit with the error. _*Segmentation Fault (core dumped)*_ i see in /var/log/messages,/var/log/asterisk/messages but all is clear no sign of any error message or warning, what does its mean its my configs problem or something wrong with asterisk i use Latest CVS. Can i use Realtime odbc instead of Mysql . extconfig.conf [settings] ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; sipfriends => mysql,asterisk,sip_friends res_mysql.conf [general] dbhost = localhost.localdoamin/127.0.0.1 dbname = asterisk dbuser = asterisk dbpass = 123456 dbport = 3306 dbsock = /var/lib/mysql/mysql.sock ___ 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] Segmentation Fault Problem
Hi, What do you think that the problem might be if a program has a segmentation fault at the same library call? The library call is from libpthread.so.0 and the call itself is "pthread_mutex_locl ( )". I have enclosed the core dump information below. The program comes up and then does the segmentation fault. (gdb) bt #0 0x40035944 in pthread_mutex_lock () from /lib/tls/libpthread.so.0 #1 0x419ede1e in register_verify (p=3D0x8163890, sin=3D0x41bfc0d0, req=3D0x41bfc0e0, uri=3D0x41bfc2fd "sip:10.0.0.5", ignore=3D0) at chan_sip.c:5032 #2 0x419f4495 in handle_request (p=3D0x8163890, req=3D0x41bfc0e0, sin=3D0x41bfc0d0, recount=3D0x0, nounlock=3D0x41bfbf48) at chan_sip.c:7861 #3 0x419f6194 in sipsock_read (id=3D0x81413f8, fd=3D20, events=3D1, ignore=3D0x0) at chan_sip.c:7959 #4 0x0805317c in ast_io_wait (ioc=3D0x813fe08, howlong=3D0) at io.c:267 #5 0x419e27db in do_monitor (data=3D0x0) at chan_sip.c:8106 #6 0x400347f3 in start_thread () from /lib/tls/libpthread.so.0 #7 0x401a462a in clone () from /lib/tls/libc.so.6 (gdb) x/5i $eip 0x40035944 : mov0xc(%esi),%ecx 0x40035947 : cmp$0x1,%ecx 0x4003594a : je 0x40035974 0x4003594c : jg 0x4003598e 0x4003594e : xor%eax,%eax (gdb) info registers eax0x0 0 ecx0x0 0 edx0x0 0 ebx0x4003dff4 1073995764 esp0x41bf01f4 0x41bf01f4 ebp0x41bf0208 0x41bf0208 esi0x0 0 edi0x32da 13018 eip0x40035944 0x40035944 eflags 0x210212 2163218 cs 0x73 115 ss 0x7b 123 ds 0x7b 123 es 0x7b 123 fs 0x0 0 gs 0x33 51 (gdb) info threads 15 process 13005 0xe410 in ?? () 14 process 13007 0xe410 in ?? () 13 process 13008 0xe410 in ?? () 12 process 13009 0xe410 in ?? () 11 process 13010 0xe410 in ?? () 10 process 13012 0xe410 in ?? () 9 process 13013 0xe410 in ?? () 8 process 13014 0xe410 in ?? () 7 process 13015 0xe410 in ?? () 6 process 13016 0xe410 in ?? () 5 process 13017 0xe410 in ?? () 4 process 13019 0xe410 in ?? () 3 process 13020 0xe410 in ?? () 2 process 13021 0xe410 in ?? () * 1 process 13018 0x40035944 in pthread_mutex_lock () from /lib/tls/libpthread.so.0 asterisk:~ # cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 8 model name : Pentium III (Coppermine) stepping: 10 cpu MHz : 996.859 cache size : 256 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 sep mtrr pge mca cmov pat pse36 mmx fxsr sse bogomips: 1978.36 Thanks, Charles ___ 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] Segmentation fault on SIP inbound
Hi, I can dial out with SIP, but any inbound call causes a segmentation fault. Before recompiling asterisk, the segfault was preceded by a "Ouch.. cannot write to audio file" error message. Here are my settings/logs. Any help is greatly appreciated... [sipgate] type=friend username=#myUSERID# host=sipgate.de fromuser=#myUSERID# fromdomain=sipgate.de nat=no context=from-sip canreinvite=no [from-sip] exten => _.,1,Wait(5) exten => _.,2,Answer exten => _.,3,Voicemail,s100 exten => _.,4,Hangup INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Record-Route: Max-Forwards: 9 Record-Route: Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKa96e.98b17d44.0 Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKa96e.8d222c41.0 Via: SIP/2.0/UDP 217.10.66.11:5060;branch=z9hG4bK687d609c From: "0" ;tag=as4eca35df To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 09 Nov 2004 11:47:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 343 Sipgate-Authentication: accepted v=0 o=root 9974 9974 IN IP4 217.10.66.11 s=session c=IN IP4 217.10.79.9 t=0 0 m=audio 55426 RTP/AVP 8 0 3 10 97 18 2 5 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:10 L16/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:5 DVI4/8000 a=direction:active a=nortpproxy:yes 18 headers, 16 lines Using latest request as basis request Sending to 217.10.79.9 : 5060 (non-NAT) Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 10 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 5 Peer audio RTP is at port 217.10.79.9:55426 Found description format PCMA Found description format PCMU Found description format GSM Found description format L16 Found description format iLBC Found description format G729 Found description format G726-32 Found description format DVI4 Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x57e(GSM|ULAW|ALAW|G726|ADPCM|SLINR|G729A|ILBC)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) Found peer 'sipgate' Segmentation fault debian:~# _ Messenger 6. 1 - lassen Sie alle Bürokollegen an Ihren genialen Ideen teilhaben. http://messenger.msn.at?DI=1031&XAPID=2532 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Segmentation fault res_features.so
On Thu, 2004-11-04 at 04:57 +0100, Serge wrote: > Have anyone some idea ? > Asterisk - latest cvs, > RedHat9 > == > Asterisk Dynamic Loader Starting: > == Parsing '/etc/asterisk/modules.conf': Found > [chan_modem.so] => (Generic Voice Modem Driver) > == Parsing '/etc/asterisk/modem.conf': Found > == Loading modem driver chan_modem_aopen.so => (A/Open (Rockwell Chipset) > ITU-2 VoiceModem Driver) > == Registered channel type 'Modem' (Generic Voice Modem Channel Driver) > [res_adsi.so] => (ADSI Resource) > == Parsing '/etc/asterisk/adsi.conf': Found > [res_features.so] => (Call Parking Resource) > == Parsing '/etc/asterisk/features.conf': Found > -- Registered extension context 'parkedcalls' > Segmentation fault > Start a new thread please. It is easier to click the mailing list address and create a clean new message than it is to click reply and delete all the contents. You also will be behaving as you should and draw no wrath from those of us who use threaded mail readers. Also, a back trace is important for a segfault so we know where in the code base that actually segfaulted. It very well may be the module that loads right after res_features.so and it faulted before it could let you know it was loading. To get a back trace, you need to get the code to drop a core file. Most linux distros turn off the core dump facility to keep you from littering the file system with the core files and you not knowing what to do with them. Use "ulimit 0", see the bash man page for more info. Then you will need to run asterisk and let it fault. Then you will have the core file. Check the gdb man page for the order of options, but you need to reference YOUR asterisk binary and the core file. Once the symbols are loaded, use the "bt" command. Post that output for someone else to help you. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation fault res_features.so
Have anyone some idea ? Asterisk - latest cvs, RedHat9 == Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_modem.so] => (Generic Voice Modem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_aopen.so => (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) == Registered channel type 'Modem' (Generic Voice Modem Channel Driver) [res_adsi.so] => (ADSI Resource) == Parsing '/etc/asterisk/adsi.conf': Found [res_features.so] => (Call Parking Resource) == Parsing '/etc/asterisk/features.conf': Found -- Registered extension context 'parkedcalls' Segmentation fault Thanks. Serge. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation Fault TDM22B & TDM04B
Hi all, i have installed two digium cards on my asterisk box a TDM04B & TDM22B. The channels are configured as show below: Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01)Channel 02: FXO Kewlstart (Default) (Slaves: 02)Channel 03: FXS Kewlstart (Default) (Slaves: 03)Channel 04: FXS Kewlstart (Default) (Slaves: 04)Channel 05: FXS Kewlstart (Default) (Slaves: 05)Channel 06: FXS Kewlstart (Default) (Slaves: 06)Channel 07: FXS Kewlstart (Default) (Slaves: 07)Channel 08: FXS Kewlstart (Default) (Slaves: 08) 8 channels configured. When i load the cards everything its fine, and the status of both cards is OK. The leds are green except the two FXO ports. The problem is when i try to load asterisk appears a segmentation fault, here is the error: "Sep 21 10:40:47 WARNING[1074404032]: chan_zap.c:7658 setup_zap: Ignoring faxdetectSep 21 10:40:47 WARNING[1074404032]: chan_zap.c:665 zt_open: Unable to specify channel 1: No such deviceSep 21 10:40:47 ERROR[1074404032]: chan_zap.c:5340 mkintf: Unable to open channel 1: No such devicehere = 0, tmp->channel = 1, channel = 1Sep 21 10:40:47 ERROR[1074404032]: chan_zap.c:7377 setup_zap: Unable to register channel '1-2'Sep 21 10:40:47 WARNING[1074404032]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap'Segmentation fault" Thanks for your help. Carlos Andres Medina Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish.___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] segmentation fault on asterisk startup
On Tuesday 13 July 2004 08:55, [EMAIL PROTECTED] wrote: > You missed my point. I'm talking about how it does data handling with > various loops and memcpys etc. I don't care about the sound quality, nor do > I care about how well written it is, I'm just making the observation based > on my previous experience based on previous auditing of software. My apologies; I read "code" as "codec" and though you were making analysis of the code by listening to the codec. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] segmentation fault on asterisk startup
On Tue, Jul 13, 2004 at 08:35:17AM -0400, Andrew Kohlsmith wrote: > On Tuesday 13 July 2004 08:22, [EMAIL PROTECTED] wrote: > > Ack, I don't like the iLBC code for the quick 3 minutes or so I looked at > > it, but it wouldn't surprise me if it was overwriting more than it should > > be on the stack. > > Why wouldn't it surprise you? I have a PRI and have 10 or 12 iLBC codecs > running during peak times. I don't understand how you can get from "I don't > like the sound of iLBC" to "iLBC must be written poorly". > You missed my point. I'm talking about how it does data handling with various loops and memcpys etc. I don't care about the sound quality, nor do I care about how well written it is, I'm just making the observation based on my previous experience based on previous auditing of software. There are a lot of variables in use with various #define'd values, (can you be sure that there is no off-by-one's, compiler eccentricities, etc?). ( it's abit like how asterisk doesn't srand() correctly, which can allow an attacker to predict what challenges someone is going to see and compute them in advance via MIT or that IAX2 should drop the idea of plaintext passwords due to forced downgrades.). That said, what processor series couldn't divide properly? if it was the cryix range / version part, it could be a problem. Thanks, Andrew Griffiths ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] segmentation fault on asterisk startup
On Tuesday 13 July 2004 08:22, [EMAIL PROTECTED] wrote: > Ack, I don't like the iLBC code for the quick 3 minutes or so I looked at > it, but it wouldn't surprise me if it was overwriting more than it should > be on the stack. Why wouldn't it surprise you? I have a PRI and have 10 or 12 iLBC codecs running during peak times. I don't understand how you can get from "I don't like the sound of iLBC" to "iLBC must be written poorly". -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] segmentation fault on asterisk startup
Hmm, block is allocated near the top of the stack. Ack, I don't like the iLBC code for the quick 3 minutes or so I looked at it, but it wouldn't surprise me if it was overwriting more than it should be on the stack. Well, I'll hand this off to the developers / people who want to spend longer looking over the code (I just happen to be handy with a debugger occasionally). > #0 0x3aeb in ?? () > #1 0x405e2752 in iLBC_encode (bytes=0x810fda0 "? ??\017`\022\021?G\"\214", >block=0xb47c, iLBCenc_inst=0x810e868) at iLBC_encode.c:93 > #2 0x405e0eea in lintoilbc_frameout (tmp=0x810e868) at codec_ilbc.c:196 > #3 0x0805ca2f in calc_cost (t=0x405e9240) at translate.c:238 > #4 0x0805ce4a in ast_register_translator (t=0x405e9240) at translate.c:299 > #5 0x405e0fef in load_module () at codec_ilbc.c:263 > #6 0x080551ce in ast_load_resource (resource_name=0x80defdb > "codec_ilbc.so") >at loader.c:312 > #7 0x08055636 in load_modules () at loader.c:407 > #8 0x08084136 in main (argc=2, argv=0xbe04) at asterisk.c:1485 > (gdb) x/5i $eip > 0x3aeb: Cannot access memory at address 0x3aeb ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] segmentation fault on asterisk startup
On Tue, Jul 13, 2004 at 01:24:18PM +0200, Andreas 'TheChaos' Groll wrote:
> [EMAIL PROTECTED] wrote:
>
> >Start off with running ulimit -c unlimited before you start asterisk. Once
> >it crashes, type gdb /path/to/asterisk core
> >
> >>From there, enter the following:
> >
> >bt
> >x/5i $eip
> >info registers
> >info threads
> >
> >and quit out. After doing that, you might want to save the output of
> >uname -a
> >cat /proc/cpuinfo
> >
> >and send it to the list.
> >
> vendor_id : CyrixInstead
> model name : 6x86MX 2.5x Core/Bus Clock
> cpu MHz : 166.405
> flags : fpu de tsc msr cx8 pge cmov mmx cyrix_arr
> bogomips: 331.77
Is anyone else running asterisk with iLBC without problems on cyrix chips?
IIRC, they where meant to be a cheaper version, so initially it made me think
that it might of been gcc emmitting a bad instruction for that cpu.
> of course I think you wanted the gdb output, I hope that's correct:
Looks good :)
>
> (gdb) bt
> #0 0x3aeb in ?? ()
> #1 0x405e2752 in iLBC_encode (bytes=0x810fda0 "? ??\017`\022\021?G\"\214",
>block=0xb47c, iLBCenc_inst=0x810e868) at iLBC_encode.c:93
> #2 0x405e0eea in lintoilbc_frameout (tmp=0x810e868) at codec_ilbc.c:196
> #3 0x0805ca2f in calc_cost (t=0x405e9240) at translate.c:238
> #4 0x0805ce4a in ast_register_translator (t=0x405e9240) at translate.c:299
> #5 0x405e0fef in load_module () at codec_ilbc.c:263
> #6 0x080551ce in ast_load_resource (resource_name=0x80defdb
> "codec_ilbc.so")
>at loader.c:312
> #7 0x08055636 in load_modules () at loader.c:407
> #8 0x08084136 in main (argc=2, argv=0xbe04) at asterisk.c:1485
> (gdb) x/5i $eip
> 0x3aeb: Cannot access memory at address 0x3aeb
Hmmm, looks like saved EIP got overwritten at some stage. I'm not familar with
the translation code, but it might be possible that its buffer was exceeded,
based upon seeing the iLBC_encode passed with a parameter on the stack. I
don't have the code handy at the moment, after I grab it I'll have
a look over it and reply to this message.
> (gdb) info registers
> eax0xbfffd924 -1073751772
> ecx0xbfffd974 -1073751692
> edx0x3 3
> ebx0x4001e89c 1073866908
> esp0xbfffd450 0xbfffd450
> ebp0xbfffd99c 0xbfffd99c
> esi0x4012819c 1074954652
> edi0x40231a9d 1076042397
> eip0x3aeb 0x3aeb
> eflags 0x10282 66178
> cs 0x23 35
> ss 0x2b 43
> ds 0x2b 43
> es 0x2b 43
> fs 0x2b 43
> gs 0x2b 43
> fctrl 0x37f895
> fstat 0x122290
> ftag 0x 65535
> fiseg 0x23 35
> fioff 0x405e4895 1079920789
> foseg 0x2b 43
> fooff 0xbfffd920 -1073751776
> fop0x11c284
> xmm0 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
>-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
> xmm1 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
>-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
> xmm2 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
>-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
> xmm3 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
>-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
> xmm4 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
>-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
> xmm5 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
>-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
> xmm6 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
>-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
> xmm7 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
>-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
> mxcsr 0x1f80 8064
> orig_eax 0x -1
> (gdb) info threads
> * 1 process 8318 0x3aeb in ?? ()
>
> Perhaps it is important to mention, that I got the bad modules again
> from a friend. His modules work within my asterisk with no errors.
> Very confusing for me. I still hope you can help.
>
> Thanks
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Re: [Asterisk-Users] segmentation fault on asterisk startup
[EMAIL PROTECTED] wrote:
Start off with running ulimit -c unlimited before you start asterisk. Once it
crashes, type gdb /path/to/asterisk core
From there, enter the following:
bt
x/5i $eip
info registers
info threads
and quit out. After doing that, you might want to save the output of
uname -a
cat /proc/cpuinfo
and send it to the list.
(Note for other people/developers, perhaps something similar to samba's panic
action might be useful, which automates a lot of this stuff might come in
use.)
Hope this helps,
Andrew Griffiths
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Ok I did.
uname -a gave me this:
Linux chaospbx 2.4.18-bf2.4 #1 Son Apr 14 09:53:28 CEST 2002 i686 unknown
cat /proc/cpuinfo:
processor : 0
vendor_id : CyrixInstead
cpu family : 6
model : 1
model name : 6x86MX 2.5x Core/Bus Clock
stepping: 4
cpu MHz : 166.405
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: yes
fpu : yes
fpu_exception : yes
cpuid level : 1
wp : yes
flags : fpu de tsc msr cx8 pge cmov mmx cyrix_arr
bogomips: 331.77
of course I think you wanted the gdb output, I hope that's correct:
(gdb) bt
#0 0x3aeb in ?? ()
#1 0x405e2752 in iLBC_encode (bytes=0x810fda0 "ÿ ÿú\017`\022\021¢G\"\214",
block=0xb47c, iLBCenc_inst=0x810e868) at iLBC_encode.c:93
#2 0x405e0eea in lintoilbc_frameout (tmp=0x810e868) at codec_ilbc.c:196
#3 0x0805ca2f in calc_cost (t=0x405e9240) at translate.c:238
#4 0x0805ce4a in ast_register_translator (t=0x405e9240) at translate.c:299
#5 0x405e0fef in load_module () at codec_ilbc.c:263
#6 0x080551ce in ast_load_resource (resource_name=0x80defdb
"codec_ilbc.so")
at loader.c:312
#7 0x08055636 in load_modules () at loader.c:407
#8 0x08084136 in main (argc=2, argv=0xbe04) at asterisk.c:1485
(gdb) x/5i $eip
0x3aeb: Cannot access memory at address 0x3aeb
(gdb) info registers
eax0xbfffd924 -1073751772
ecx0xbfffd974 -1073751692
edx0x3 3
ebx0x4001e89c 1073866908
esp0xbfffd450 0xbfffd450
ebp0xbfffd99c 0xbfffd99c
esi0x4012819c 1074954652
edi0x40231a9d 1076042397
eip0x3aeb 0x3aeb
eflags 0x10282 66178
cs 0x23 35
ss 0x2b 43
ds 0x2b 43
es 0x2b 43
fs 0x2b 43
gs 0x2b 43
fctrl 0x37f895
fstat 0x122290
ftag 0x 65535
fiseg 0x23 35
fioff 0x405e4895 1079920789
foseg 0x2b 43
fooff 0xbfffd920 -1073751776
fop0x11c284
xmm0 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm1 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm2 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm3 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm4 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm5 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm6 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm7 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
mxcsr 0x1f80 8064
orig_eax 0x -1
(gdb) info threads
* 1 process 8318 0x3aeb in ?? ()
Perhaps it is important to mention, that I got the bad modules again
from a friend. His modules work within my asterisk with no errors.
Very confusing for me. I still hope you can help.
Thanks
signature.asc
Description: OpenPGP digital signature
Re: [Asterisk-Users] segmentation fault on asterisk startup
> > Ok, just removed this last module works, asterisk is starting without > errors anymore, but I wanted to use ILBC codec so it's importan for me. > > Can anyone help me, getting this to work? > Start off with running ulimit -c unlimited before you start asterisk. Once it crashes, type gdb /path/to/asterisk core >From there, enter the following: bt x/5i $eip info registers info threads and quit out. After doing that, you might want to save the output of uname -a cat /proc/cpuinfo and send it to the list. (Note for other people/developers, perhaps something similar to samba's panic action might be useful, which automates a lot of this stuff might come in use.) Hope this helps, Andrew Griffiths ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] segmentation fault on asterisk startup
Hi, I write to this list, because I didn't find anything on the net. I installed asterisk using bristuff-0.0.2 without any errors, but when I start asterisk with "asterisk -vvvc" I get following error: [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ILBC to SLINR, cost 127 Segmentation fault Removing codec_ilbc.so from /usr/lib/asterisk/modules shows up the next error: [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder) == Registered translator 'lpc10tolin' from format LPC10 to SLINR, cost 63 Segmentation fault Ok, just removed this last module works, asterisk is starting without errors anymore, but I wanted to use ILBC codec so it's importan for me. Can anyone help me, getting this to work? I'm running Debian 3.0 (2.4.18-bf2.4) with bristuff-0.0.2 and the zaphfc module loaded. Thanks for any replies. Bye Andreas signature.asc Description: OpenPGP digital signature
[Asterisk-Users] Segmentation fault, exit status 139, ...
Hi!
I am running an * 0.7.2 on an X86 debian stable 2.4.25 (with
backports.org). The HW I am using is Digium's E100P on an HP DL 380.
Quite often it crashes, e.g. after a call has finished. Below some logs
form the * Console as well as from the /var/log/asterisk/messages
(Replaced some stuff with XXX).
Any idea what there could be the reason for this segmentaion fault?
What other indormation (e.g. configs) would be required to analyse
this problem further?
Thanx for you help!
cheers,
Bernie
* Console:
Apr 5 18:01:18 WARNING[24594]: app_dial.c:331 wait_for_answer:
Unable to forward voice
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
-- Executing Dial("SIP/xxx.switch.ch-0894ea58",
"Zap/g1/04176XXX") in new stack
-- Called g1/04176
-- Zap/1-1 is ringing
-- Hungup 'Zap/1-1'
== Spawn extension (SIP, +4176XXX, 1) exited non-zero on
'SIP/xxx.switch.ch-0894ea58'
astra*CLI>
/usr/sbin/safe_asterisk: line 6: 20873 Segmentation fault
asterisk ${ASTARGS} 1>&/dev/${TTY} &/dev/${TTY} &/dev/${TTY} http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)
Sorry about that: bt gives the following output: #0 0x401519fc in mallopt () from /lib/i686/libc.so.6 #1 0x40150c61 in malloc () from /lib/i686/libc.so.6 #2 0x40157dd0 in strdup () from /lib/i686/libc.so.6 #3 0x0805603b in cfg_process (tmp=0x80ea890, _tmpc=0x47a6a26c, _last=0x47a6a270, buf=0x65747865 , lineno=183, configfile=0x458d083d "extensions.conf", includelevel=0) at config.c:57 #4 0x08055ac2 in __ast_load (configfile=0x458d083d "extensions.conf", tmp=0x80ea890, _tmpc=0x47a6a26c, _last=0x47a6a270, includelevel=0) at config.c:731 #5 0x08055edd in ast_load (configfile=0x8 ) at config.c:766 #6 0x458cf815 in pbx_load_module () at pbx_config.c:1543 #7 0x458ccdaf in reload () at pbx_config.c:1683 #8 0x08055372 in ast_module_reload () at loader.c:159 #9 0x0806b8ba in handle_reload (fd=44, argc=1, argv=0x47a6a5fc) at cli.c:105 #10 0x0806b6aa in ast_cli_command (fd=44, s=0x8 ) at cli.c:1006 #11 0x08083c80 in netconsole (vconsole=0x80b9c28) at asterisk.c:192 #12 0x40021811 in pthread_start_thread () from /lib/i686/libpthread.so.0 I see extensions.conf mentioned a couple of times. Could this be caused by a configuration error? --- Steven Critchfield <[EMAIL PROTECTED]> wrote: > On Wed, 2003-09-24 at 08:41, jerk face wrote: > > Ok, here is the real gdb output. > > > > This GDB was configured as > > "i586-mandrake-linux-gnu"... > > Core was generated by `asterisk'. > > Program terminated with signal 11, Segmentation > fault. > > ... > > ... > > ... > > > > Loaded symbols for > > /usr/lib/asterisk/modules/cdr_csv.so > > Reading symbols from > > /usr/lib/asterisk/modules/app_setcidnum.so...done. > > Loaded symbols for > > /usr/lib/asterisk/modules/app_setcidnum.so > > #0 0x401519fc in mallopt () from > /lib/i686/libc.so.6 > > > > Ok .. so what does this mean? > > > > Thank you in advance. > > > This is where you type 'bt' and find out how it made > it to that frame. > > > > --- Martin Pycko <[EMAIL PROTECTED]> wrote: > > > actually > > > > > > gdb /usr/sbin/asterisk core.6044, sorry > > > > > > On Tue, 23 Sep 2003, jerk face wrote: > > > > > > > I keep getting segmentation faults when I do a > > > reload. > > > > > > > > Here are the core file outputs from gdb: > > > > (I have three of them and they produce the > same > > > > output) > > > > > > > > (gdb) core core.6044 > > > > Core was generated by `asterisk'. > > > > Program terminated with signal 11, > Segmentation > > > fault. > > > > #0 0x401519fc in ?? () > > > > > > > > > > > > I have no idea what that means, but if > somebody > > > could > > > > point me in the right direction, that would be > > > great. > > > > > > > > Thank you for your time. > > > > > > > > __ > > > > Do you Yahoo!? > > > > Yahoo! SiteBuilder - Free, easy-to-use web > site > > > design software > > > > http://sitebuilder.yahoo.com > > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PROTECTED] > > > > > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > __ > > Do you Yahoo!? > > Yahoo! SiteBuilder - Free, easy-to-use web site > design software > > http://sitebuilder.yahoo.com > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)
I am running Mandrake 9.1 if that makes a difference. --- Patrick <[EMAIL PROTECTED]> wrote: > On Wed, 2003-09-24 at 15:41, jerk face wrote: > > Ok, here is the real gdb output. > > > > This GDB was configured as > > "i586-mandrake-linux-gnu"... > > Core was generated by `asterisk'. > > Program terminated with signal 11, Segmentation > fault. > > ... > > ... > > ... > > > > Loaded symbols for > > /usr/lib/asterisk/modules/cdr_csv.so > > Reading symbols from > > /usr/lib/asterisk/modules/app_setcidnum.so...done. > > Loaded symbols for > > /usr/lib/asterisk/modules/app_setcidnum.so > > #0 0x401519fc in mallopt () from > /lib/i686/libc.so.6 > > > > Ok .. so what does this mean? > > > > Thank you in advance. > > > > I started experiencing segfaults a while back too. > Both after a reload > and during startup when loading res_adsi.so. What > fixed it for me was > changing -O6 in all Makefiles to -O2. I'm using Red > Hat 9 and it has > been said that Red Hat does some funky stuff with > their gcc so that may > be causing it to segfault when compiled with -O6. > > Regards, > Patrick > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)
On Wed, 2003-09-24 at 08:41, jerk face wrote: > Ok, here is the real gdb output. > > This GDB was configured as > "i586-mandrake-linux-gnu"... > Core was generated by `asterisk'. > Program terminated with signal 11, Segmentation fault. > ... > ... > ... > > Loaded symbols for > /usr/lib/asterisk/modules/cdr_csv.so > Reading symbols from > /usr/lib/asterisk/modules/app_setcidnum.so...done. > Loaded symbols for > /usr/lib/asterisk/modules/app_setcidnum.so > #0 0x401519fc in mallopt () from /lib/i686/libc.so.6 > > Ok .. so what does this mean? > > Thank you in advance. This is where you type 'bt' and find out how it made it to that frame. > --- Martin Pycko <[EMAIL PROTECTED]> wrote: > > actually > > > > gdb /usr/sbin/asterisk core.6044, sorry > > > > On Tue, 23 Sep 2003, jerk face wrote: > > > > > I keep getting segmentation faults when I do a > > reload. > > > > > > Here are the core file outputs from gdb: > > > (I have three of them and they produce the same > > > output) > > > > > > (gdb) core core.6044 > > > Core was generated by `asterisk'. > > > Program terminated with signal 11, Segmentation > > fault. > > > #0 0x401519fc in ?? () > > > > > > > > > I have no idea what that means, but if somebody > > could > > > point me in the right direction, that would be > > great. > > > > > > Thank you for your time. > > > > > > __ > > > Do you Yahoo!? > > > Yahoo! SiteBuilder - Free, easy-to-use web site > > design software > > > http://sitebuilder.yahoo.com > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > __ > Do you Yahoo!? > Yahoo! SiteBuilder - Free, easy-to-use web site design software > http://sitebuilder.yahoo.com > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)
On Wed, 2003-09-24 at 15:41, jerk face wrote: > Ok, here is the real gdb output. > > This GDB was configured as > "i586-mandrake-linux-gnu"... > Core was generated by `asterisk'. > Program terminated with signal 11, Segmentation fault. > ... > ... > ... > > Loaded symbols for > /usr/lib/asterisk/modules/cdr_csv.so > Reading symbols from > /usr/lib/asterisk/modules/app_setcidnum.so...done. > Loaded symbols for > /usr/lib/asterisk/modules/app_setcidnum.so > #0 0x401519fc in mallopt () from /lib/i686/libc.so.6 > > Ok .. so what does this mean? > > Thank you in advance. > I started experiencing segfaults a while back too. Both after a reload and during startup when loading res_adsi.so. What fixed it for me was changing -O6 in all Makefiles to -O2. I'm using Red Hat 9 and it has been said that Red Hat does some funky stuff with their gcc so that may be causing it to segfault when compiled with -O6. Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)
Ok, here is the real gdb output. This GDB was configured as "i586-mandrake-linux-gnu"... Core was generated by `asterisk'. Program terminated with signal 11, Segmentation fault. ... ... ... Loaded symbols for /usr/lib/asterisk/modules/cdr_csv.so Reading symbols from /usr/lib/asterisk/modules/app_setcidnum.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_setcidnum.so #0 0x401519fc in mallopt () from /lib/i686/libc.so.6 Ok .. so what does this mean? Thank you in advance. --- Martin Pycko <[EMAIL PROTECTED]> wrote: > actually > > gdb /usr/sbin/asterisk core.6044, sorry > > On Tue, 23 Sep 2003, jerk face wrote: > > > I keep getting segmentation faults when I do a > reload. > > > > Here are the core file outputs from gdb: > > (I have three of them and they produce the same > > output) > > > > (gdb) core core.6044 > > Core was generated by `asterisk'. > > Program terminated with signal 11, Segmentation > fault. > > #0 0x401519fc in ?? () > > > > > > I have no idea what that means, but if somebody > could > > point me in the right direction, that would be > great. > > > > Thank you for your time. > > > > __ > > Do you Yahoo!? > > Yahoo! SiteBuilder - Free, easy-to-use web site > design software > > http://sitebuilder.yahoo.com > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 23 September 2003 19:04, jerk face wrote: > I keep getting segmentation faults when I do a reload. Do what Critchfield and Pycko told you to do, but I'm betting you're using chan_h323 with a gatekeeper (it's a known bug I think). That's what's killing my Asterisk on reload: (gdb) bt #0 0x40571b3e in PSocket::os_select(int, fd_set*, fd_set*, fd_set*, PScalarArray const&, PTimeInterval const&) () from /usr/lib/libpt_linux_x86_r.so.1.5.2 #1 0x4056aef2 in PSocket::Select(PSocket::SelectList&, PSocket::SelectList&, PSocket::SelectList&, PTimeInterval const&) () from /usr/lib/libpt_linux_x86_r.so.1.5.2 #2 0x4056a8c5 in PSocket::Select(PSocket::SelectList&, PTimeInterval const&) () from /usr/lib/libpt_linux_x86_r.so.1.5.2 #3 0x40c380e7 in H323TransportUDP::DiscoverGatekeeper(H323Gatekeeper&, H323RasPDU&, H323TransportAddress const&) () from /usr/lib/libh323_linux_x86_r.so.1.12.2 #4 0x40c48ebf in H323Gatekeeper::StartDiscovery(H323TransportAddress const&) () from /usr/lib/libh323_linux_x86_r.so.1.12.2 #5 0x40c48cfc in H323Gatekeeper::DiscoverByAddress(H323TransportAddress const&) () from /usr/lib/libh323_linux_x86_r.so.1.12.2 #6 0x40bfd59c in H323EndPoint::SetGatekeeper(PString const&, H323Transport*) () from /usr/lib/libh323_linux_x86_r.so.1.12.2 #7 0x403ac4d4 in h323_set_gk (gatekeeper_discover=0, gatekeeper=0x403b4dc0 "195.135.216.2", secret=0x403b4e40 "") at ast_h323.cpp:1005 #8 0x403a5231 in reload () at chan_h323.c:1643 #9 0x080554c2 in ast_module_reload () at loader.c:159 #10 0x0806d10a in handle_reload (fd=153, argc=1, argv=0xbd9ff61c) at cli.c:105 #11 0x0806cefa in ast_cli_command (fd=153, s=0x0) at cli.c:1006 #12 0x08085770 in netconsole (vconsole=0x80bcc28) at asterisk.c:193 #13 0x40023463 in pthread_detach () from /lib/libpthread.so.0 - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/cWlx2TEAILET3McRAhjnAKCCJI93ty4OxpGZwIMNlchRaWiiTwCeNk/Y HVqmHi76DIZsofmMF46Bp1w= =gIC7 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)
On Tue, 2003-09-23 at 12:04, jerk face wrote: > I keep getting segmentation faults when I do a reload. > > Here are the core file outputs from gdb: > (I have three of them and they produce the same > output) > > (gdb) core core.6044 you need to run gdb like follows gdb core.6044 `which asterisk` This lets gdb load the asterisk binary with all the symbols in it to understand the core file. Then if you could issue the command 'bt' inside of gdb it will give a nice listing of functions called to get to the point the crash happened. > Core was generated by `asterisk'. > Program terminated with signal 11, Segmentation fault. > #0 0x401519fc in ?? () > > > I have no idea what that means, but if somebody could > point me in the right direction, that would be great. > > Thank you for your time. > > __ > Do you Yahoo!? > Yahoo! SiteBuilder - Free, easy-to-use web site design software > http://sitebuilder.yahoo.com > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)
actually gdb /usr/sbin/asterisk core.6044, sorry On Tue, 23 Sep 2003, jerk face wrote: > I keep getting segmentation faults when I do a reload. > > Here are the core file outputs from gdb: > (I have three of them and they produce the same > output) > > (gdb) core core.6044 > Core was generated by `asterisk'. > Program terminated with signal 11, Segmentation fault. > #0 0x401519fc in ?? () > > > I have no idea what that means, but if somebody could > point me in the right direction, that would be great. > > Thank you for your time. > > __ > Do you Yahoo!? > Yahoo! SiteBuilder - Free, easy-to-use web site design software > http://sitebuilder.yahoo.com > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)
gdb /usr/src/asterisk core.6044 then 'bt' Martin On Tue, 23 Sep 2003, jerk face wrote: > I keep getting segmentation faults when I do a reload. > > Here are the core file outputs from gdb: > (I have three of them and they produce the same > output) > > (gdb) core core.6044 > Core was generated by `asterisk'. > Program terminated with signal 11, Segmentation fault. > #0 0x401519fc in ?? () > > > I have no idea what that means, but if somebody could > point me in the right direction, that would be great. > > Thank you for your time. > > __ > Do you Yahoo!? > Yahoo! SiteBuilder - Free, easy-to-use web site design software > http://sitebuilder.yahoo.com > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation Fault on reload (gdb output included)
I keep getting segmentation faults when I do a reload. Here are the core file outputs from gdb: (I have three of them and they produce the same output) (gdb) core core.6044 Core was generated by `asterisk'. Program terminated with signal 11, Segmentation fault. #0 0x401519fc in ?? () I have no idea what that means, but if somebody could point me in the right direction, that would be great. Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2
what did the trick for me is: 1. Use codec statements: allow=SOMECODEC) 2. bind * to an IP instead of "bind=0.0.0.0" (bind=IP_OF_YOUR_*_BOX) Ta Senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of mattf Sent: 11 September 2003 21:30 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 Hello, Don't know if this is related but I just got a segmentation fault today while trying to register my new SNOM200 phone: *CLI> *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '"Matt Florell" ' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '"Matt Florell" ' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4651 (handle_request): Failed to authenticate user ;tag=3g5l65mbnk for SUBSCRIBE NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '"Matt Florell" ' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '"Matt Florell" ' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '"Matt Florell" ' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '"Matt Florell" ' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4651 (handle_request): Failed to authenticate user ;tag=3g5l65mbnk for SUBSCRIBE Segmentation fault then I started asterisk back up and it registered and then gave me another error: *CLI> *CLI> -- Registered SIP 'mattf2' at 10.10.10.14 port 5060 expires 3600 *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '"Matt Florell" ' failed for '10.10.10.14' Any idea what causes this? could I have possibly just configured the Snom incorrectly to cause this? Thanks, MATT--- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, September 11, 2003 4:01 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 Is there any chance that you could send a tcpdump of the system tying to make a connection? Open up two terminal sessions to your server. In one, type "tcpdump -w foo". In the other session, start asterisk and let it sit there hanging for a few minutes. Stop Ctrl-C in the terminal session running tcpdump and send me the file "foo" that was created by tcpdump. You might want to gzip it if it's large and send it to me off list. -Original Message- From: Senad Jordanovic [mailto:[EMAIL PROTECTED] Sent: Thursday, September 11, 2003 2:48 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 Tim, thanks for your answer. I tried, all of the options you suggested, and still the same... * hangs. It is interesting that it does the same think, when it tries to register my FWD account and there are no problems with registering IAX account with NuFone. I can make iconnect/fwd/iax calls with no major problems. Any other suggestions? Senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 11 September 2003 20:14 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 I assume that from your previous post that you are using iconnect Is your register line in the format: Register => 18005551212:[EMAIL PROTECTED]/18005551212 I've had good luck using the IP address vs. the fully qualified hostname. Remember that the register line goes in the [general] section of sip.conf. Also, are you using the latest CVS release of *? -Original Message- From: Senad Jordanovic [mailto:[EMAIL PROTECTED] Sent: Thursday, September 11, 2003 1:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 I am really desperate to have any help on this problem below as it prevents us from making any further progress. Is there anyone out there who can help? Thanks Senad - Hi, Allowing registration to iconnect by using "register => XXX" in sip.conf file does not start *, but instead produces "segmentation fault" and * hangs. Commenting out "register => " allows * to start wit
RE: [Asterisk-Users] Segmentation fault due to SIP registration N UMBER 2
Hello, Don't know if this is related but I just got a segmentation fault today while trying to register my new SNOM200 phone: *CLI> *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '"Matt Florell" ' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '"Matt Florell" ' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4651 (handle_request): Failed to authenticate user ;tag=3g5l65mbnk for SUBSCRIBE NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '"Matt Florell" ' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '"Matt Florell" ' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '"Matt Florell" ' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '"Matt Florell" ' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4651 (handle_request): Failed to authenticate user ;tag=3g5l65mbnk for SUBSCRIBE Segmentation fault then I started asterisk back up and it registered and then gave me another error: *CLI> *CLI> -- Registered SIP 'mattf2' at 10.10.10.14 port 5060 expires 3600 *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '"Matt Florell" ' failed for '10.10.10.14' Any idea what causes this? could I have possibly just configured the Snom incorrectly to cause this? Thanks, MATT--- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, September 11, 2003 4:01 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 Is there any chance that you could send a tcpdump of the system tying to make a connection? Open up two terminal sessions to your server. In one, type "tcpdump -w foo". In the other session, start asterisk and let it sit there hanging for a few minutes. Stop Ctrl-C in the terminal session running tcpdump and send me the file "foo" that was created by tcpdump. You might want to gzip it if it's large and send it to me off list. -Original Message- From: Senad Jordanovic [mailto:[EMAIL PROTECTED] Sent: Thursday, September 11, 2003 2:48 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 Tim, thanks for your answer. I tried, all of the options you suggested, and still the same... * hangs. It is interesting that it does the same think, when it tries to register my FWD account and there are no problems with registering IAX account with NuFone. I can make iconnect/fwd/iax calls with no major problems. Any other suggestions? Senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 11 September 2003 20:14 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 I assume that from your previous post that you are using iconnect Is your register line in the format: Register => 18005551212:[EMAIL PROTECTED]/18005551212 I've had good luck using the IP address vs. the fully qualified hostname. Remember that the register line goes in the [general] section of sip.conf. Also, are you using the latest CVS release of *? -Original Message- From: Senad Jordanovic [mailto:[EMAIL PROTECTED] Sent: Thursday, September 11, 2003 1:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 I am really desperate to have any help on this problem below as it prevents us from making any further progress. Is there anyone out there who can help? Thanks Senad - Hi, Allowing registration to iconnect by using "register => XXX" in sip.conf file does not start *, but instead produces "segmentation fault" and * hangs. Commenting out "register => " allows * to start with no problems. Anyone know why is this happening? -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mai
RE: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2
Is there any chance that you could send a tcpdump of the system tying to make a connection? Open up two terminal sessions to your server. In one, type "tcpdump -w foo". In the other session, start asterisk and let it sit there hanging for a few minutes. Stop Ctrl-C in the terminal session running tcpdump and send me the file "foo" that was created by tcpdump. You might want to gzip it if it's large and send it to me off list. -Original Message- From: Senad Jordanovic [mailto:[EMAIL PROTECTED] Sent: Thursday, September 11, 2003 2:48 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 Tim, thanks for your answer. I tried, all of the options you suggested, and still the same... * hangs. It is interesting that it does the same think, when it tries to register my FWD account and there are no problems with registering IAX account with NuFone. I can make iconnect/fwd/iax calls with no major problems. Any other suggestions? Senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 11 September 2003 20:14 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 I assume that from your previous post that you are using iconnect Is your register line in the format: Register => 18005551212:[EMAIL PROTECTED]/18005551212 I've had good luck using the IP address vs. the fully qualified hostname. Remember that the register line goes in the [general] section of sip.conf. Also, are you using the latest CVS release of *? -Original Message- From: Senad Jordanovic [mailto:[EMAIL PROTECTED] Sent: Thursday, September 11, 2003 1:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 I am really desperate to have any help on this problem below as it prevents us from making any further progress. Is there anyone out there who can help? Thanks Senad - Hi, Allowing registration to iconnect by using "register => XXX" in sip.conf file does not start *, but instead produces "segmentation fault" and * hangs. Commenting out "register => " allows * to start with no problems. Anyone know why is this happening? -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2
Tim, thanks for your answer. I tried, all of the options you suggested, and still the same... * hangs. It is interesting that it does the same think, when it tries to register my FWD account and there are no problems with registering IAX account with NuFone. I can make iconnect/fwd/iax calls with no major problems. Any other suggestions? Senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 11 September 2003 20:14 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 I assume that from your previous post that you are using iconnect Is your register line in the format: Register => 18005551212:[EMAIL PROTECTED]/18005551212 I've had good luck using the IP address vs. the fully qualified hostname. Remember that the register line goes in the [general] section of sip.conf. Also, are you using the latest CVS release of *? -Original Message- From: Senad Jordanovic [mailto:[EMAIL PROTECTED] Sent: Thursday, September 11, 2003 1:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 I am really desperate to have any help on this problem below as it prevents us from making any further progress. Is there anyone out there who can help? Thanks Senad - Hi, Allowing registration to iconnect by using "register => XXX" in sip.conf file does not start *, but instead produces "segmentation fault" and * hangs. Commenting out "register => " allows * to start with no problems. Anyone know why is this happening? -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2
I assume that from your previous post that you are using iconnect Is your register line in the format: Register => 18005551212:[EMAIL PROTECTED]/18005551212 I've had good luck using the IP address vs. the fully qualified hostname. Remember that the register line goes in the [general] section of sip.conf. Also, are you using the latest CVS release of *? -Original Message- From: Senad Jordanovic [mailto:[EMAIL PROTECTED] Sent: Thursday, September 11, 2003 1:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 I am really desperate to have any help on this problem below as it prevents us from making any further progress. Is there anyone out there who can help? Thanks Senad - Hi, Allowing registration to iconnect by using "register => XXX" in sip.conf file does not start *, but instead produces "segmentation fault" and * hangs. Commenting out "register => " allows * to start with no problems. Anyone know why is this happening? -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2
I am really desperate to have any help on this problem below as it prevents us from making any further progress. Is there anyone out there who can help? Thanks Senad - Hi, Allowing registration to iconnect by using "register => XXX" in sip.conf file does not start *, but instead produces "segmentation fault" and * hangs. Commenting out "register => " allows * to start with no problems. Anyone know why is this happening? -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation fault due to SIP registration
Hi, Allowing registration to iconnect by using "register => XXX" in sip.conf file does not start *, but instead produces "segmentation fault" and * hangs. Commenting out "register => " allows * to start with no problems. Anyone know why is this happening? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Albertson Sent: 10 September 2003 20:32 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Security vulnerability report > >Read the security vulnerability. It referenced CVS > as of a certain > >date. If you aren't keeping up with CVS changes, > why are you running > >CVS at all? One would hope people are not using the latest CVS checkup as their production system. Most sane people do a bit better quality control and testing then that on a misson critical system. So fielded systems are likely to be a bit in back of CVS. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation fault
arrr, errr
try Dial(phone/phone0|5)
the quicknet card is definately NOT a zap device
On Mon, 4 Aug 2003 17:29:55 -0500 (CDT), [EMAIL PROTECTED] wrote:
>When i make a call from x100p to quicknet
>
>-- Executing Dial("Zap/1-1", "Phone/phone0|5") in new stack
>Segmentation fault
>
>
>extension file
>
>[quicknet]
>exten =3D> _1XX,1,Dial(Zap/0/${EXTEN})
>
>phone file
>
>[interfaces]
>mode=3Ddialtone
>format=3Dslinear
>echocancel=3Dmedium
>context=3Dquicknet
>device =3D> /dev/phone0
>
>___
>Asterisk-Users mailing list
>[EMAIL PROTECTED]
>http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation fault
When i make a call from x100p to quicknet
-- Executing Dial("Zap/1-1", "Phone/phone0|5") in new stack
Segmentation fault
extension file
[quicknet]
exten =3D> _1XX,1,Dial(Zap/0/${EXTEN})
phone file
[interfaces]
mode=3Ddialtone
format=3Dslinear
echocancel=3Dmedium
context=3Dquicknet
device =3D> /dev/phone0
___
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[EMAIL PROTECTED]
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Re: [Asterisk-Users] segmentation fault with asterisk and OH323
It's OK. I change my oh323.conf file and I don't have segmantation fault anymore. Thanks Rattana - Original Message - From: "Rattana BIV" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, August 01, 2003 3:42 PM Subject: Re: [Asterisk-Users] segmentation fault with asterisk and OH323 > By my phone I call H323 client (Netmeeting) I can talk. > Everything is OK. > > But when I hangup I have the segmentation fault. > These are the last asterisk log before the segmentation fault: > > -- Called 192.168.1.200 > -- H323:193 answered CAPI[contr1/26]/1 > DEBUG[23569]: File channel.c, Line 2145 (ast_channel_bridge): Didn't get a > frame from channel: CAPI[contr1/26]/1 > DEBUG[23569]: File channel.c, Line 2213 (ast_channel_bridge): Bridge stops > bridging channels CAPI[contr1/26]/1 and H323:193 > > > Something strange is, when I call the phone from netmeeting it's work > perfectly. > > What do you think ? > (you can see my oh323.conf in attach file) > > > Rattana > > > - Original Message - > From: "Michael Manousos" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Friday, August 01, 2003 3:07 PM > Subject: Re: [Asterisk-Users] segmentation fault with asterisk and OH323 > > > > Rattana BIV wrote: > > > Hi, > > > > > > I got a segmantation fault When I call to computer (h323) from phone. > > > > > > I use asterisk-oh323 0.5.4 and chan_capi.0.2.2 drivers. > > > > More info (config files, screen log, backtrace of core file) is needed. > > > > > > > > Someone know where the problem ? > > > > > > > > > Regards > > > Rattana > > > > Michael. > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] segmentation fault with asterisk and OH323
By my phone I call H323 client (Netmeeting) I can talk. Everything is OK. But when I hangup I have the segmentation fault. These are the last asterisk log before the segmentation fault: -- Called 192.168.1.200 -- H323:193 answered CAPI[contr1/26]/1 DEBUG[23569]: File channel.c, Line 2145 (ast_channel_bridge): Didn't get a frame from channel: CAPI[contr1/26]/1 DEBUG[23569]: File channel.c, Line 2213 (ast_channel_bridge): Bridge stops bridging channels CAPI[contr1/26]/1 and H323:193 Something strange is, when I call the phone from netmeeting it's work perfectly. What do you think ? (you can see my oh323.conf in attach file) Rattana - Original Message - From: "Michael Manousos" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, August 01, 2003 3:07 PM Subject: Re: [Asterisk-Users] segmentation fault with asterisk and OH323 > Rattana BIV wrote: > > Hi, > > > > I got a segmantation fault When I call to computer (h323) from phone. > > > > I use asterisk-oh323 0.5.4 and chan_capi.0.2.2 drivers. > > More info (config files, screen log, backtrace of core file) is needed. > > > > > Someone know where the problem ? > > > > > > Regards > > Rattana > > Michael. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users oh323.conf Description: Binary data
Re: [Asterisk-Users] segmentation fault with asterisk and OH323
Rattana BIV wrote: Hi, I got a segmantation fault When I call to computer (h323) from phone. I use asterisk-oh323 0.5.4 and chan_capi.0.2.2 drivers. More info (config files, screen log, backtrace of core file) is needed. Someone know where the problem ? Regards Rattana Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] segmentation fault with asterisk and OH323
Hi, I got a segmantation fault When I call to computer (h323) from phone. I use asterisk-oh323 0.5.4 and chan_capi.0.2.2 drivers. Someone know where the problem ? Regards Rattana
Re: [Asterisk-Users] Segmentation fault with chan_oh323
That is another problem I hope the developers would pay attention to. ulaw codec segfaulting when it is used by h323 side of connection for both incoming and outgoing calls. At least with chan_oh323. If I set alaw codec for h323 it works fine regardless of codec on SIP side. Michael On Thursday 17 July 2003 03:36 am, Mark Thompson wrote: > This also happened to me when I was using the same codec with both oh323 > and SIP, if I forced it to alaw on oh323 and ulaw on SIP the connection > worked. I also tried h323 instead of oh323 which works okay but you have > to use earlier versions of pwlib and openh323. > Mark > > -Original Message- > From: Michael Ulitskiy [mailto:[EMAIL PROTECTED] > Sent: 16 July 2003 23:44 > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Segmentation fault with chan_oh323 > > > Hi, > > I'm trying to interconnect sip and h323 endpoints using asterisk and > asterisk crashes with segmentation fault whenever h323 > connection needs to be established. It registers with gatekeeper ok > though. Here are the symptoms. If the call initiated by SIP device, > asterisk replies to it "Trying" and then silently crashes (it launched > as asterisk -cd). In debug log I can see the following: Jul 16 > 18:11:52 DEBUG[196621]: File pbx.c, Line 1123 (pbx_extension_helper): > Launching 'Dial' Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line > 1393 (oh323_request): In oh323_request. Jul 16 18:11:52 DEBUG[196621]: > File chan_oh323.c, Line 1394 (oh323_request): type=oh323, format=4, > data=. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, > Line 1440 (oh323_request): Created new call structure 0 (2428 bytes). > That's it. If the call initiated by H323 device, then I see > *CLI> > WrapH323Connection::WrapH323Connection: WrapH323Connection created. > Segmentation fault and debug log shows: Jul 16 18:33:12 DEBUG[196621]: > File chan_oh323.c, Line 2141 (init_h323_connection): In > init_h323_connection... Jul 16 18:33:12 DEBUG[196621]: File > chan_oh323.c, Line 2180 (init_h323_connection): Created new call > structure 0 (2428 bytes). Jul 16 18:33:12 DEBUG[196621]: File > chan_oh323.c, Line 1527 (copy_call_details): --- CALL DETAILS --- Jul 16 > 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1528 > (copy_call_details): call_token = ip$192.168.0.227:5018/92 Jul 16 > 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1529 > (copy_call_details): call_source_alias = tnt [192.168.0.227] > Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1530 > (copy_call_details): call_dest_alias = 12125551234 12125551234 > ip$192.168.0.70:1720 > Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1531 > (copy_call_details): call_source_e164 = phone number Jul 16 18:33:12 > DEBUG[196621]: File chan_oh323.c, Line 1532 (copy_call_details): > call_dest_e164 = 12125551234 That's it. And gatekeeper log shows that > after normal ARQ-ACF exchange originating device immediately sent DRQ. > If anybody knows a reason for this (and the way to fix it of course ;)), > I'd appreciate if you let me know. If you need any additional info to > troubleshoot it, let me know too. Thank a lot. > > Michael > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation fault with chan_oh323
Michael Ulitskiy wrote: Hi, I'm trying to interconnect sip and h323 endpoints using asterisk and asterisk crashes with segmentation fault whenever h323 connection needs to be established. It registers with gatekeeper ok though. Here are the symptoms. If the call initiated by SIP device, asterisk replies to it "Trying" and then silently crashes (it launched as asterisk -cd). In debug log I can see the following: Jul 16 18:11:52 DEBUG[196621]: File pbx.c, Line 1123 (pbx_extension_helper): Launching 'Dial' Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1393 (oh323_request): In oh323_request. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1394 (oh323_request): type=oh323, format=4, data=. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1440 (oh323_request): Created new call structure 0 (2428 bytes). That's it. If the call initiated by H323 device, then I see *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. Segmentation fault and debug log shows: Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2141 (init_h323_connection): In init_h323_connection... Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2180 (init_h323_connection): Created new call structure 0 (2428 bytes). Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1527 (copy_call_details): --- CALL DETAILS --- Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1528 (copy_call_details): call_token = ip$192.168.0.227:5018/92 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1529 (copy_call_details): call_source_alias = tnt [192.168.0.227] Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1530 (copy_call_details): call_dest_alias = 12125551234 12125551234 ip$192.168.0.70:1720 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1531 (copy_call_details): call_source_e164 = phone number Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1532 (copy_call_details): call_dest_e164 = 12125551234 That's it. And gatekeeper log shows that after normal ARQ-ACF exchange originating device immediately sent DRQ. If anybody knows a reason for this (and the way to fix it of course ;)), I'd appreciate if you let me know. If you need any additional info to troubleshoot it, let me know too. First of all, in oh323.conf, set wrapLibTraceLevel=3 libTraceLevel=3 libTraceFile=/tmp/trace.txt Run Asterisk again, with -vvvcd, and make it crash. Then send me (offlist) the trace file, the screen log and a backtrace of the core file dumped. Thank a lot. Michael Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Segmentation fault with chan_oh323
Hi Everybody, I am new to Asterisk. Can anybody suggest me some link where I can find architecture level detail of this system. My aim is to find out how easy it is to port it on a new hardware (T1/E1 and POTS)? Any input is highly appreciated. Regards Arun -Original Message- From: Mark Thompson [mailto:[EMAIL PROTECTED] Sent: 17 July 2003 13:07 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Segmentation fault with chan_oh323 This also happened to me when I was using the same codec with both oh323 and SIP, if I forced it to alaw on oh323 and ulaw on SIP the connection worked. I also tried h323 instead of oh323 which works okay but you have to use earlier versions of pwlib and openh323. Mark -Original Message- From: Michael Ulitskiy [mailto:[EMAIL PROTECTED] Sent: 16 July 2003 23:44 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Segmentation fault with chan_oh323 Hi, I'm trying to interconnect sip and h323 endpoints using asterisk and asterisk crashes with segmentation fault whenever h323 connection needs to be established. It registers with gatekeeper ok though. Here are the symptoms. If the call initiated by SIP device, asterisk replies to it "Trying" and then silently crashes (it launched as asterisk -cd). In debug log I can see the following: Jul 16 18:11:52 DEBUG[196621]: File pbx.c, Line 1123 (pbx_extension_helper): Launching 'Dial' Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1393 (oh323_request): In oh323_request. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1394 (oh323_request): type=oh323, format=4, data=. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1440 (oh323_request): Created new call structure 0 (2428 bytes). That's it. If the call initiated by H323 device, then I see *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. Segmentation fault and debug log shows: Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2141 (init_h323_connection): In init_h323_connection... Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2180 (init_h323_connection): Created new call structure 0 (2428 bytes). Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1527 (copy_call_details): --- CALL DETAILS --- Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1528 (copy_call_details): call_token = ip$192.168.0.227:5018/92 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1529 (copy_call_details): call_source_alias = tnt [192.168.0.227] Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1530 (copy_call_details): call_dest_alias = 12125551234 12125551234 ip$192.168.0.70:1720 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1531 (copy_call_details): call_source_e164 = phone number Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1532 (copy_call_details): call_dest_e164 = 12125551234 That's it. And gatekeeper log shows that after normal ARQ-ACF exchange originating device immediately sent DRQ. If anybody knows a reason for this (and the way to fix it of course ;)), I'd appreciate if you let me know. If you need any additional info to troubleshoot it, let me know too. Thank a lot. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Segmentation fault with chan_oh323
This also happened to me when I was using the same codec with both oh323 and SIP, if I forced it to alaw on oh323 and ulaw on SIP the connection worked. I also tried h323 instead of oh323 which works okay but you have to use earlier versions of pwlib and openh323. Mark -Original Message- From: Michael Ulitskiy [mailto:[EMAIL PROTECTED] Sent: 16 July 2003 23:44 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Segmentation fault with chan_oh323 Hi, I'm trying to interconnect sip and h323 endpoints using asterisk and asterisk crashes with segmentation fault whenever h323 connection needs to be established. It registers with gatekeeper ok though. Here are the symptoms. If the call initiated by SIP device, asterisk replies to it "Trying" and then silently crashes (it launched as asterisk -cd). In debug log I can see the following: Jul 16 18:11:52 DEBUG[196621]: File pbx.c, Line 1123 (pbx_extension_helper): Launching 'Dial' Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1393 (oh323_request): In oh323_request. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1394 (oh323_request): type=oh323, format=4, data=. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1440 (oh323_request): Created new call structure 0 (2428 bytes). That's it. If the call initiated by H323 device, then I see *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. Segmentation fault and debug log shows: Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2141 (init_h323_connection): In init_h323_connection... Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2180 (init_h323_connection): Created new call structure 0 (2428 bytes). Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1527 (copy_call_details): --- CALL DETAILS --- Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1528 (copy_call_details): call_token = ip$192.168.0.227:5018/92 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1529 (copy_call_details): call_source_alias = tnt [192.168.0.227] Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1530 (copy_call_details): call_dest_alias = 12125551234 12125551234 ip$192.168.0.70:1720 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1531 (copy_call_details): call_source_e164 = phone number Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1532 (copy_call_details): call_dest_e164 = 12125551234 That's it. And gatekeeper log shows that after normal ARQ-ACF exchange originating device immediately sent DRQ. If anybody knows a reason for this (and the way to fix it of course ;)), I'd appreciate if you let me know. If you need any additional info to troubleshoot it, let me know too. Thank a lot. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation fault with chan_oh323
Hi, I'm trying to interconnect sip and h323 endpoints using asterisk and asterisk crashes with segmentation fault whenever h323 connection needs to be established. It registers with gatekeeper ok though. Here are the symptoms. If the call initiated by SIP device, asterisk replies to it "Trying" and then silently crashes (it launched as asterisk -cd). In debug log I can see the following: Jul 16 18:11:52 DEBUG[196621]: File pbx.c, Line 1123 (pbx_extension_helper): Launching 'Dial' Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1393 (oh323_request): In oh323_request. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1394 (oh323_request): type=oh323, format=4, data=. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1440 (oh323_request): Created new call structure 0 (2428 bytes). That's it. If the call initiated by H323 device, then I see *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. Segmentation fault and debug log shows: Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2141 (init_h323_connection): In init_h323_connection... Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2180 (init_h323_connection): Created new call structure 0 (2428 bytes). Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1527 (copy_call_details): --- CALL DETAILS --- Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1528 (copy_call_details): call_token = ip$192.168.0.227:5018/92 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1529 (copy_call_details): call_source_alias = tnt [192.168.0.227] Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1530 (copy_call_details): call_dest_alias = 12125551234 12125551234 ip$192.168.0.70:1720 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1531 (copy_call_details): call_source_e164 = phone number Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1532 (copy_call_details): call_dest_e164 = 12125551234 That's it. And gatekeeper log shows that after normal ARQ-ACF exchange originating device immediately sent DRQ. If anybody knows a reason for this (and the way to fix it of course ;)), I'd appreciate if you let me know. If you need any additional info to troubleshoot it, let me know too. Thank a lot. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
