[asterisk-users] Asterisk crashes when storing voicemail via odbc
Hi all, I'm working on migrating all of my servers to store voicemail in a mysql database via odbc. I've got a development server that I can reconfigure and test at will. When it's configured to store vm on the file system, it seems to be rock solid. However, when I ONLY change it to store vm in the database, it becomes very unstable. Here's what it's doing. When I attempt leave a voicemail, I am prompted to leave a message. Once I have left a message, the console locks up and I have to killall -9 to get it to restart and become responsive again. I'm running Asterisk 13.14.0 built by root @ server on a x86_64 running Linux on 2017-06-20 14:27:06 UTC For odbc, I've got unixODBC 2.3.2-r2. Are these the versions I should be using? If so, any recommendations as to how to troubleshoot this would be most welcome. TIA, -- Mike Diehl -- _ -- 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] Asterisk Crashes Randomly with Cepstral Swift TTS
Hi Bryan, On 10/18/2014 11:47 PM, Bryan Burroughs wrote: All, Has anyone seen this before? This appears to be a Swift or app_swift bug. I'm having a difficult time finding any information or support on this. I haven't used app_Swift with Cepstral but iirc it wasn't deemed very stable. Asterisk version: Asterisk 11.6-cert4 built by asterisk @ ivrd02 on a x86_64 running Linux on 2014-08-11 13:55:25 UTC OS: Linux livrp03 2.6.32-431.11.2.el6.x86_64 #1 SMP Mon Mar 3 13:32:45 EST 2014 x86_64 x86_64 x86_64 GNU/Linux If you are not tied to the certified Asterisk version then perhaps try using the latest Asterisk version (currently 11.13.0). When Asterisk crashes, the backtrace always looks something like the following: [snip] The out of bounds line looks like it may be pointing to the issue. *argv = {0x0, 0xb9b0 Address 0xb9b0 out of bounds, 0x0}* Have you tried contacting the app_swift developer and/or filed a bug at https://issues.asterisk.org/jira/secure/Dashboard.jspa ? Should I look into using another TTS engine? You could try UniMRCP which sits between Asterisk and Cepstral replacing app_swift: http://unimrcp.org HTH, Patrick -- _ -- 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] Asterisk Crashes Randomly with Cepstral Swift TTS
Do you or anyone have any experience using Swift with UniMRCP? I tried a few months ago but gave up. The documentation is not very clear. Yes, I am reaching out to the developer of app_swift. thanks, Bryan Burroughs On 10/19/2014 08:02 AM, Patrick Laimbock wrote: Hi Bryan, On 10/18/2014 11:47 PM, Bryan Burroughs wrote: All, Has anyone seen this before? This appears to be a Swift or app_swift bug. I'm having a difficult time finding any information or support on this. I haven't used app_Swift with Cepstral but iirc it wasn't deemed very stable. Asterisk version: Asterisk 11.6-cert4 built by asterisk @ ivrd02 on a x86_64 running Linux on 2014-08-11 13:55:25 UTC OS: Linux livrp03 2.6.32-431.11.2.el6.x86_64 #1 SMP Mon Mar 3 13:32:45 EST 2014 x86_64 x86_64 x86_64 GNU/Linux If you are not tied to the certified Asterisk version then perhaps try using the latest Asterisk version (currently 11.13.0). When Asterisk crashes, the backtrace always looks something like the following: [snip] The out of bounds line looks like it may be pointing to the issue. *argv = {0x0, 0xb9b0 Address 0xb9b0 out of bounds, 0x0}* Have you tried contacting the app_swift developer and/or filed a bug at https://issues.asterisk.org/jira/secure/Dashboard.jspa ? Should I look into using another TTS engine? You could try UniMRCP which sits between Asterisk and Cepstral replacing app_swift: http://unimrcp.org HTH, Patrick -- _ -- 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] Asterisk Crashes Randomly with Cepstral Swift TTS
All, Has anyone seen this before? This appears to be a Swift or app_swift bug. I'm having a difficult time finding any information or support on this. Asterisk version: Asterisk 11.6-cert4 built by asterisk @ ivrd02 on a x86_64 running Linux on 2014-08-11 13:55:25 UTC OS: Linux livrp03 2.6.32-431.11.2.el6.x86_64 #1 SMP Mon Mar 3 13:32:45 EST 2014 x86_64 x86_64 x86_64 GNU/Linux When Asterisk crashes, the backtrace always looks something like the following: [Thread debugging using libthread_db enabled] Core was generated by `/opt/asterisk/sbin/asterisk -f -C /opt/asterisk/etc/asterisk/asterisk.c'. Program terminated with signal 11, Segmentation fault. #0 0x7f79439ba061 in LM_set_port_number () from /opt/swift/lib/libswift.so.6 #0 0x7f79439ba061 in LM_set_port_number () from /opt/swift/lib/libswift.so.6 No symbol table info available. #1 0x7f79439bb7c6 in LM_init () from /opt/swift/lib/libswift.so.6 No symbol table info available. #2 0x7f79439c5f0d in swift_engine_open () from /opt/swift/lib/libswift.so.6 No symbol table info available. #3 0x7f7943bf9b6e in app_exec (chan=0x7f7914009e08, data=0x7f7928f9c680 413 E 3RD ST) at app_swift.c:370 res = 0 max_digits = 0 timeout = 0 alreadyran = 0 ms = 0 len = 6043056 availatend = 0 argv = {0x0, 0xb9b0 Address 0xb9b0 out of bounds, 0x0} text = 0x7f7928f9a230 413 E 3RD ST rc = 0x0 tmp_exten = \000 results = '\000' repeats 19 times u = 0x7f7974001dc0 f = 0x7f7928f9a4c0 next = {tv_sec = 6014649, tv_usec = 21020457123253} ps = 0x7f79740165e0 parse = 0x7f7928f9a230 413 E 3RD ST old_writeformat = {id = 687449040, fattr = {format_attr = {32633, 5184330, 0, 687448768, 32633, 5710841, 0, 48, 48, 687448992, 32633, 687448800, 32633, 1946162544, 0, 1946162544, 32633, 8448136, 0, 84, 0, 4294967292, 0, 687449104, 32633, 5827993, 0, 84, 0, 1717986919, 1717986918, 8448135, 0, 5623124, 0, 687449040, 32633, 5630801, 0, 9, 0, 687449040, 0, 687449104, 32633, 40, 0, 1413666990, 0, 5222777, 0, 1946239199, 32633, 687449192, 32633, 80, 0, 0, 0, 2, 0, 5187840, 0, 1946207160}, rtp_marker_bit = 121 'y'}} args = {argc = 1, argv = 0x7f7928f9a280, text = 0x7f7928f9a230 413 E 3RD ST, timeout = 0x0, max_digits = 0x0} myf = {f = {frametype = 0, subclass = {integer = 0, format = {id = 0, fattr = {format_attr = {0, 0, 0, 0, 257, 0, 2, 0, 1839600224, 32767, 645521943, 1069709714, 4, 0, 0, 0, 1839600224, 32767, 687448144, 32633, 127525399, 3225199326, 3012944407, 1069709582, 1, 0 repeats 33 times, 1839600016, 32767, 2, 0, 34150224, 0}, rtp_marker_bit = 0 '\000'}}}, datalen = 0, samples = 97009376, mallocd = 0, mallocd_hdr_len = 177, offset = 1, src = 0x7f7928f9a050 \207, incomplete sequence \350\200, data = {ptr = 0x0, uint32 = 0, pad = \000\000\000\000\000\000\000}, delivery = {tv_sec = 0, tv_usec = 0}, frame_list = {next = 0x0}, flags = 0, ts = 0, len = 0, seqno = 0}, offset = \000\000\000\000\000\000\000\000\310\314\335\005\000\000\000\000\310\314\335\005, '\000' repeats 20 times, `\177\316\005\000\000\000\000\377\377\000\000\001\000\000\000\000\000\000\000\000\000\000 The out of bounds line looks like it may be pointing to the issue. *argv = {0x0, 0xb9b0 Address 0xb9b0 out of bounds, 0x0}* Should I look into using another TTS engine? thanks, -- Bryan Burroughs -- _ -- 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] Asterisk crashes when reloading configs...
I am having a very strange problem. We use Asterisk 11.X (have tried several versions, including certified) which reads its config files in realtime from a SQLITE3 database. Everything runs fine but lately asterisk has been crashing when we issue a reload command via Manager. Our web interface uses AMI to reload the dialplan and right after it does that ( I can see the results on the CLI) asterisk crashes. This does not seem to happen every time but some days it crashes often. Any ideas where to start looking for the problem? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 -- _ -- 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] Asterisk crashes when reloading configs...
On Wed, Jul 2, 2014 at 12:38 PM, Carlos Chavez cur...@telecomabmex.com wrote: I am having a very strange problem. We use Asterisk 11.X (have tried several versions, including certified) which reads its config files in realtime from a SQLITE3 database. Everything runs fine but lately asterisk has been crashing when we issue a reload command via Manager. Our web interface uses AMI to reload the dialplan and right after it does that ( I can see the results on the CLI) asterisk crashes. This does not seem to happen every time but some days it crashes often. Any ideas where to start looking for the problem? Please get a backtrace illustrating the problem: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace Once you have a properly generated backtrace, open an issue on issues.asterisk.org. Thanks - Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Asterisk crashes suddenly
Hello friends, I have been experienced suddenly stops for my Asterisk server, I do not why is it happening. Asterisk's debug messages only tell me I have lacked g729 codec for translation to one peer minutes before the crashes occur [2014-05-27 09:48:30] WARNING[15384][C-017c] channel.c: Unable to find a codec translation path from (ulaw) to (g729) [2014-05-27 09:48:30] WARNING[15384][C-017c] chan_sip.c: Asked to transmit frame type g729, while native formats is (ulaw) read/write = ulaw/ulaw [2014-05-27 09:48:30] WARNING[15384][C-017c] chan_sip.c: Asked to transmit frame type ulaw, while native formats is (g729) read/write = ulaw/slin [2014-05-27 09:48:30] WARNING[15384][C-017c] channel.c: Codec mismatch on channel SIP/20108-0051 setting write format to g729 from ulaw native formats (ulaw) And it stops after a failed attended transfer between two of my SIP peers. [2014-05-27 09:48:32] WARNING[15384][C-017c] channel.c: No path to translate from SIP/20108-0051 to SIP/30201-0052 [2014-05-27 09:48:32] WARNING[15384][C-017c] channel.c: Can't make SIP/20108-0051 and SIP/30201-0052 compatible [2014-05-27 09:48:32] WARNING[15384][C-017c] features.c: Bridge failed on channels SIP/20108-0051 and SIP/30201-0052 There are no more messages from Asterisk but I have found this message on kern.log, messages and syslog.1 on /var/log/: May 27 09:48:32 pbx-thor-PE kernel: [334427.888524] asterisk[15384] general protection ip:482a13 sp:7f335b87c898 error:0 in asterisk[40+221000] I am using Debian 7.5 64 bits with Asterisk 11.9.0 Thank you! Elder D. Arohuanca Lima - Peru -- _ -- 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] Asterisk crashes suddenly
Quoting Daniel - Asterisk (earohua...@gmail.com): And it stops after a failed attended transfer between two of my SIP peers. May 27 09:48:32 pbx-thor-PE kernel: [334427.888524] asterisk[15384] general protection ip:482a13 sp:7f335b87c898 error:0 in asterisk[40+221000] I am using Debian 7.5 64 bits with Asterisk 11.9.0 Have you checked for core dumps? AFAIK, Debian packages create core dumps on segfaults in /tmp(?). You could use that to further pin point the issue, perhaps. $ gdb /usr/sbin/asterisk /tmp/something.core gdb bt full If it is reproducable, try recompiling Asterisk with debug symbols to get a clearer backtrace of what happened. -Sndr. -- | Do not meet girl in park, instead park meet in girl! | 4096R/20CC6CD2 - 6D40 1A20 B9AA 87D4 84C7 FBD6 F3A9 9442 20CC 6CD2 -- _ -- 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] Asterisk crashes at meetme kick all
Thanks a lot Patrick. Regards Rajib Deka Siemens Ltd. -- Message: 7 Date: Mon, 17 Feb 2014 10:22:02 +0100 From: Patrick Laimbock patr...@laimbock.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk crashes at meetme kick all Message-ID: 5301d4ba.3040...@laimbock.com Content-Type: text/plain; charset=windows-1252; format=flowed On 17-02-14 09:26, Deka, Rajib IN MAA SL wrote: Dear Forum, I have encountered a similar issue as below in Asterisk 10.0.0. Asterisk crashed while executing ?meetme kick all? CLI command from manager interface. The link says the issue has been closed however I am not able to identify in which release of asterisk this issue has been fixed. Please help. _https://issues.asterisk.org/jira/browse/ASTERISK-15741_ AFAICT this issue has not been fixed due to inactivity. Note the Suspended due to lack of activity remark. Also the 1.6 version mentioned in the bugreport is EOL. Version 10.0.0 you mentioned is also EOL so any bugreport you file against version 10.0.0 will not be acted upon unless you can reproduce it with the latest Asterisk version 11.x.x or 12.x.x. I recommend you upgrade to an Asterisk LTS (long term support) version like the latest 11.x.x (currently 11.7.0). For more information about Asterisk LTS versions go to: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions If you still see a bug when running Asterisk 11.x.x (or 12.x.x) you can report it at the Asterisk issue tracker at: https://issues.asterisk.org/jira/secure/Dashboard.jspa Before filing a bug please read the information at: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines#AsteriskIssueGuidelines-Howtoreportabug -- Patrick -- -- _ -- 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] Asterisk crashes at meetme kick all
Dear Forum, I have encountered a similar issue as below in Asterisk 10.0.0. Asterisk crashed while executing meetme kick all CLI command from manager interface. The link says the issue has been closed however I am not able to identify in which release of asterisk this issue has been fixed. Please help. https://issues.asterisk.org/jira/browse/ASTERISK-15741 With best regards, Rajib Deka Siemens Ltd. -- _ -- 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] Asterisk crashes at meetme kick all
On 17-02-14 09:26, Deka, Rajib IN MAA SL wrote: Dear Forum, I have encountered a similar issue as below in Asterisk 10.0.0. Asterisk crashed while executing “meetme kick all” CLI command from manager interface. The link says the issue has been closed however I am not able to identify in which release of asterisk this issue has been fixed. Please help. _https://issues.asterisk.org/jira/browse/ASTERISK-15741_ AFAICT this issue has not been fixed due to inactivity. Note the Suspended due to lack of activity remark. Also the 1.6 version mentioned in the bugreport is EOL. Version 10.0.0 you mentioned is also EOL so any bugreport you file against version 10.0.0 will not be acted upon unless you can reproduce it with the latest Asterisk version 11.x.x or 12.x.x. I recommend you upgrade to an Asterisk LTS (long term support) version like the latest 11.x.x (currently 11.7.0). For more information about Asterisk LTS versions go to: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions If you still see a bug when running Asterisk 11.x.x (or 12.x.x) you can report it at the Asterisk issue tracker at: https://issues.asterisk.org/jira/secure/Dashboard.jspa Before filing a bug please read the information at: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines#AsteriskIssueGuidelines-Howtoreportabug -- Patrick -- _ -- 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] Asterisk crashes on high IO load
Did you take a look at /var/log/syslog /var/log/asterisk/messages ? Using Debian? Take a look at iotop (apt-get install iotop). There you can see information about which process consumes high io load. Am 04.04.2011 17:23, schrieb Maximilian Grobecker: Hello Thorsten, the system has 4 GB RAM and about 2,5 GB free so swap space is not used or exhausted. Maybe the high load is not cause of this crashes but it's the only thing the crashes can be reproduced with. Thank you! Maximilian Grobecker Am 04.04.2011 16:03, schrieb Thorsten Göllner: Take a look with top at your system when high io load is seen. Maybe the machine is running out of ram and starts swapping? Am 04.04.2011 15:04, schrieb Maximilian Grobecker: Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker -- _ -- 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 -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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] Asterisk crashes on high IO load
Hi, the log files contained (sometimes) lines about refcount -1 in astobj.c. I also generated core dumps and analyzed them - but there were always errors in another module. Mabye I found the solution: Asterisk seems to crash when a required module cannot be loaded fast enough due to heavy disk usage. When I move the modules directory to another hard disk Asterisk runs fine. I'm using autoload=yes in modules.conf and have several noload lines in it. Is there a possibility to say asterisk to load all modules to RAM at start time and not on demand? Thanks and greetings Max Am 05.04.2011 09:45, schrieb Thorsten Göllner: Did you take a look at /var/log/syslog /var/log/asterisk/messages ? Using Debian? Take a look at iotop (apt-get install iotop). There you can see information about which process consumes high io load. Am 04.04.2011 17:23, schrieb Maximilian Grobecker: Hello Thorsten, the system has 4 GB RAM and about 2,5 GB free so swap space is not used or exhausted. Maybe the high load is not cause of this crashes but it's the only thing the crashes can be reproduced with. Thank you! Maximilian Grobecker Am 04.04.2011 16:03, schrieb Thorsten Göllner: Take a look with top at your system when high io load is seen. Maybe the machine is running out of ram and starts swapping? Am 04.04.2011 15:04, schrieb Maximilian Grobecker: Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker -- _ -- 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 -- _ -- 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] Asterisk crashes on high IO load
On 6/04/11 12:39 AM, Maximilian Grobecker wrote: Hi, the log files contained (sometimes) lines about refcount -1 in astobj.c. I also generated core dumps and analyzed them - but there were always errors in another module. Mabye I found the solution: Asterisk seems to crash when a required module cannot be loaded fast enough due to heavy disk usage. When I move the modules directory to another hard disk Asterisk runs fine. I'm using autoload=yes in modules.conf and have several noload lines in it. Is there a possibility to say asterisk to load all modules to RAM at start time and not on demand? You could compile Asterisk with embedded modules? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] Asterisk crashes on high IO load
Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker -- _ -- 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] Asterisk crashes on high IO load
Take a look with top at your system when high io load is seen. Maybe the machine is running out of ram and starts swapping? Am 04.04.2011 15:04, schrieb Maximilian Grobecker: Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker -- _ -- 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 -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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] Asterisk crashes on high IO load
Hello Thorsten, the system has 4 GB RAM and about 2,5 GB free so swap space is not used or exhausted. Maybe the high load is not cause of this crashes but it's the only thing the crashes can be reproduced with. Thank you! Maximilian Grobecker Am 04.04.2011 16:03, schrieb Thorsten Göllner: Take a look with top at your system when high io load is seen. Maybe the machine is running out of ram and starts swapping? Am 04.04.2011 15:04, schrieb Maximilian Grobecker: Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker -- _ -- 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] Asterisk crashes on high IO load
Are you using IAX? There are some problems causing crashes for us related to laggyness on IAX channels with 1.8 versions. There are a bunch of problems with IAX related to https://issues.asterisk.org/view.php?id=17521 Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Maximilian Grobecker Sent: 04 April 2011 16:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk crashes on high IO load Hello Thorsten, the system has 4 GB RAM and about 2,5 GB free so swap space is not used or exhausted. Maybe the high load is not cause of this crashes but it's the only thing the crashes can be reproduced with. Thank you! Maximilian Grobecker Am 04.04.2011 16:03, schrieb Thorsten Göllner: Take a look with top at your system when high io load is seen. Maybe the machine is running out of ram and starts swapping? Am 04.04.2011 15:04, schrieb Maximilian Grobecker: Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker -- _ -- 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 -- _ -- 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] Asterisk Crashes - Segmentation Fault
Hi Dan, I was trying to make the Invite working. I am getting following error when i try to make a call. [Aug 8 16:55:22] NOTICE[15082]: chan_local.c:534 local_call: No such extension/context 73...@default while calling Local channel [Aug 8 16:55:22] NOTICE[15082]: channel.c:4042 __ast_request_and_dial: Unable to call channel Local/73281 [Aug 8 16:55:22] ERROR[12166]: pbx.c:9301 device_state_cb: Received invalid event that had no device IE [Aug 8 16:55:22] ERROR[12166]: app_queue.c:1099 device_state_cb: Received invalid event that had no device IE Following is my dialplan in /etc/asterisk/extensions.conf [outgoing] exten = _73...,1,Dial(SIP/callman02SIP/callman01/${EXTEN:2}) exten = _73...,n,Congestion following is in lib/defines.php //Outcall defaults define (CHAN_TYPE, Local); //Use Local to let dialplan decide which chan define (OUT_CONTEXT, outgoing); //Select a context to place the call from define (OUT_PEER, ); // Use this if not using CHAN_TYPE Local define (OUT_CALL_CID, Parlez 1996); // Caller ID for Invites --Manmohan Singh On Fri, Aug 6, 2010 at 12:46 AM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I commented locale.php in defines.php and it perfectly worked well. Now i am wondering what is this invite participants for, while adding conference. wherein it asks for first name, lastname, emailaddress telephone number.. The 'Invite Others' option is mostly for installs that do not have a consistent e-mail environment, and are using the SERVER mailer. This feature lets the server send invite emails to multiple parties. In my environments we have Exchange and Outlook, so I prefer the CLIENT mailer, and I can manage the invitations in my mail client Let me brief you how i had done this setup. I had created a SIP trunk between Cisco Call manager and Asterisk server. And i am using webmeetme for Audio conferencing. Sounds familiar. I put this package together after wasting too much money and time trying to make an expensive Cisco conferencing solution work. Other than the invite participants, while the conf call is going on we get couple of more options, when we click to the current ongoing conference number. End call -- To end the conference call Yes Extend -- I am sure this is to extend the time of the call for which its scheduled for, but not sure on how much time does it extends by default OR is there any way to define the customized time on whatever required. 10 minutes is the default. I thought I had made it configurable in lib/defines.php, but no I have it hard coded in conf_add (to be fixed in the next release now). You can search for +600 and change it to any value you like. Invite-- When i click this button it asks me telephone number. I assume this is any number which asterisk server can reach as per the dialplan configured in extension.conf in /etc/asterisk.. Though this invite button looks pretty much interesting to use but whenever i enter any phone number it says System error not sure if am understand this wrongly. You understand it correctly, but the default settings are likely not working. Check out the section 'Outcall defaults' in lib/defines.php. It is likely you need to change the OUT_CONTEXT at a minimum. Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] Asterisk Crashes - Segmentation Fault
Hi Dan, I commented locale.php in defines.php and it perfectly worked well. Now i am wondering what is this invite participants for, while adding conference. wherein it asks for first name, lastname, emailaddress telephone number.. Let me brief you how i had done this setup. I had created a SIP trunk between Cisco Call manager and Asterisk server. And i am using webmeetme for Audio conferencing. Other than the invite participants, while the conf call is going on we get couple of more options, when we click to the current ongoing conference number. End call -- To end the conference call Extend -- I am sure this is to extend the time of the call for which its scheduled for, but not sure on how much time does it extends by default OR is there any way to define the customized time on whatever required. Invite-- When i click this button it asks me telephone number. I assume this is any number which asterisk server can reach as per the dialplan configured in extension.conf in /etc/asterisk.. Though this invite button looks pretty much interesting to use but whenever i enter any phone number it says System error not sure if am understand this wrongly. Please correct me if i am wrong. --Manmohan Singh On Wed, Aug 4, 2010 at 9:14 PM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I had tried the new version of webmeetme i.e., 4.0.2 The recording works very well. Great! I see following php errors whenever i try to add in conference. [Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice: Undefined variable: order in /var/www/html/web-meetme/meetme_control.php on line 278, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4 You can ignore the Notices. They are fairly harmless, and only mean that variable is not set by the code or being passed in on the URL. You can turn off notices in /etc/php.ini if they bother you. Also the Reports link doesnt display anything and in httpd error logs it gives me following php errors: [Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning: include(locale.php) [a href='function.include'function.include/a]: failed to open stream: No such file or directory in /var/www/html/web-meetme/lib/defines.php on line 3, referer: http://10.1.1.30/web-meetme/daily.php? In lib/defines.php, either comment out the 3rd line or add ../ before locale.php- include(../locale.php); But that is not likely why you do not get the reports. The most likely cause is A PHP notice is being thrown while the GD code is rendering the graph, resulting in a corrupt image which your browser cannot display. Check these settings /etc/php.ini- error_reporting = E_ALL display_errors = Off Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] Asterisk Crashes - Segmentation Fault
Manmohan wrote: I commented locale.php in defines.php and it perfectly worked well. Now i am wondering what is this invite participants for, while adding conference. wherein it asks for first name, lastname, emailaddress telephone number.. The 'Invite Others' option is mostly for installs that do not have a consistent e-mail environment, and are using the SERVER mailer. This feature lets the server send invite emails to multiple parties. In my environments we have Exchange and Outlook, so I prefer the CLIENT mailer, and I can manage the invitations in my mail client Let me brief you how i had done this setup. I had created a SIP trunk between Cisco Call manager and Asterisk server. And i am using webmeetme for Audio conferencing. Sounds familiar. I put this package together after wasting too much money and time trying to make an expensive Cisco conferencing solution work. Other than the invite participants, while the conf call is going on we get couple of more options, when we click to the current ongoing conference number. End call -- To end the conference call Yes Extend -- I am sure this is to extend the time of the call for which its scheduled for, but not sure on how much time does it extends by default OR is there any way to define the customized time on whatever required. 10 minutes is the default. I thought I had made it configurable in lib/defines.php, but no I have it hard coded in conf_add (to be fixed in the next release now). You can search for +600 and change it to any value you like. Invite-- When i click this button it asks me telephone number. I assume this is any number which asterisk server can reach as per the dialplan configured in extension.conf in /etc/asterisk.. Though this invite button looks pretty much interesting to use but whenever i enter any phone number it says System error not sure if am understand this wrongly. You understand it correctly, but the default settings are likely not working. Check out the section 'Outcall defaults' in lib/defines.php. It is likely you need to change the OUT_CONTEXT at a minimum. Dan -- _ -- 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] Asterisk Crashes - Segmentation Fault
Hi Dan, I had tried the new version of webmeetme i.e., 4.0.2 The recording works very well. I see following php errors whenever i try to add in conference. [Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice: Undefined variable: order in /var/www/html/web-meetme/meetme_control.php on line 278, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4 [Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice: Undefined variable: sens in /var/www/html/web-meetme/meetme_control.php on line 278, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4 [Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice: Undefined variable: current_page in /var/www/html/web-meetme/meetme_control.php on line 278, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4 [Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice: Undefined variable: dateReq in /var/www/html/web-meetme/meetme_control.php on line 573, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4 Also the Reports link doesnt display anything and in httpd error logs it gives me following php errors: [Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning: include(locale.php) [a href='function.include'function.include/a]: failed to open stream: No such file or directory in /var/www/html/web-meetme/lib/defines.php on line 3, referer: http://10.1.1.30/web-meetme/daily.php? [Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning: include() [a href='function.include'function.include/a]: Failed opening 'locale.php' for inclusion (include_path='.:/usr/share/pear:/usr/share/php') in /var/www/html/web-meetme/lib/defines.php on line 3, referer: http://10.1.1.30/web-meetme/daily.php? [Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning: include(locale.php) [a href='function.include'function.include/a]: failed to open stream: No such file or directory in /var/www/html/web-meetme/lib/email_body.php on line 3, referer: http://10.1.1.30/web-meetme/daily.php? [Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning: include() [a href='function.include'function.include/a]: Failed opening 'locale.php' for inclusion (include_path='.:/usr/share/pear:/usr/share/php') in /var/www/html/web-meetme/lib/email_body.php on line 3, referer: http://10.1.1.30/web-meetme/daily.php? Otherwise i am able to record and play the recorded file from the speaker button. --Manmohan Singh On Fri, Jul 30, 2010 at 9:10 PM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: There was on very silly mistake and i missed to check that properly. Really apologize for that. Following change was done to get the conf-recording into the proper path: chown -R asterisk:asterisk /var/lib/asterisk/sounds/conf-recordings following is the output: [r...@linuxtest sounds]# ll total 6416 drwxrwxr-x 2 asterisk asterisk4096 Jul 30 08:29 conf-recordings [r...@linuxtest sounds]# ll conf-recordings/ total 4060 -rw-r--r-- 1 asterisk asterisk 4150124 Jul 30 08:27 meetme-conf-rec-74438-1280463795.8.wav The only thing now is no speaker icon onto the webpage when i click to past conference link. The web interface cannot find the recording. The reason it cannot is that the name is wrong. By wrong, I mean it contains information that the database and program is not aware of (1280463795.8). To make this clear, if this conference was the 3rd one you ever scheduled on this system the correct file name would be- meetme-conf-rec-74438-3.wav using the format meetme-conf-rec-%PIN%-%BOOKID%.wav The database knows the pin and bookid, so it can construct the file name and test if it exists. Do you say that if i shift from 4.0.1 to 4.0.2 will fix the issue (of getting speaker icon in past conference)? I was not able to get the change into app_meetme to use the bookid in the filename, even though it has access to bookid. I gave up and now store the filename in the database, which app_meetme will use if it exists. Other that a handful of bug-fixes, this is the major difference between 4.0.1 and 4.0.2 Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] Asterisk Crashes - Segmentation Fault
Manmohan wrote: I had tried the new version of webmeetme i.e., 4.0.2 The recording works very well. Great! I see following php errors whenever i try to add in conference. [Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice: Undefined variable: order in /var/www/html/web-meetme/meetme_control.php on line 278, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4 You can ignore the Notices. They are fairly harmless, and only mean that variable is not set by the code or being passed in on the URL. You can turn off notices in /etc/php.ini if they bother you. Also the Reports link doesnt display anything and in httpd error logs it gives me following php errors: [Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning: include(locale.php) [a href='function.include'function.include/a]: failed to open stream: No such file or directory in /var/www/html/web-meetme/lib/defines.php on line 3, referer: http://10.1.1.30/web-meetme/daily.php? In lib/defines.php, either comment out the 3rd line or add ../ before locale.php- include(../locale.php); But that is not likely why you do not get the reports. The most likely cause is A PHP notice is being thrown while the GD code is rendering the graph, resulting in a corrupt image which your browser cannot display. Check these settings /etc/php.ini- error_reporting = E_ALL display_errors = Off Dan -- _ -- 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] Asterisk Crashes - Segmentation Fault
Manmohan wrote: I did added the record option in user options as well. $Mod_Options = array(array(_(Announce), I), array(_(Record), r)); $User_Options = array(array(_(Announce), I), array(_(Listen Only), m), array(_(Wait for Leader), w), array(_(Record), r)); And the gre8 news is, it did worked this time. But it saved the recorded file in the following path: That is good to hear. /var/lib/asterisk/sounds/ with the name as meetme-conf-rec-74438-1280463795.8.wav Than i tried to move the file to /var/lib/asterisk/sounds/conf-recordings/ just to see that it gives me a speaker icon when i click to past conferences. Unfortunately i couldnt see this speaker icon to hear this recorded conference .wav file. I am not surprised. By default MeetMe creates unique file names by appending pin-uniqueid, but uniqueid is not know until the conference starts, so the web interface does not know what to look for. Part of the changes to app_meetme included setting the realtime filename to use. I tried to download the .wav file into my windows machine and the filed played well.. like i mentioned in my earlier mail that following line i had added in lib/define.php, please correct me if i am wrong: define (RECORDING_PATH, /var/lib/asterisk/sounds/conf-recordings/); Do you think This recording path is taking the effect here? That setting effect where the WMM interface looks for recordings and not where Asterisk puts them. Looking back at your email history, I see you are on 4.0.1. After all of the suggestions, I remembered that I too found problems with recordings and addressed them in 4.0.2 That version adds a field to the database and stores the recording names in the database. I recommend using that version instead of 4.0.1. You can move your copy of lib/defines.php to the 4.0.2 install and keep your changes. Dan -- _ -- 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] Asterisk Crashes - Segmentation Fault
Hi Dan, There was on very silly mistake and i missed to check that properly. Really apologize for that. Following change was done to get the conf-recording into the proper path: chown -R asterisk:asterisk /var/lib/asterisk/sounds/conf-recordings following is the output: [r...@linuxtest sounds]# ll total 6416 drwxrwxr-x 2 asterisk asterisk4096 Jul 30 08:29 conf-recordings [r...@linuxtest sounds]# ll conf-recordings/ total 4060 -rw-r--r-- 1 asterisk asterisk 4150124 Jul 30 08:27 meetme-conf-rec-74438-1280463795.8.wav The only thing now is no speaker icon onto the webpage when i click to past conference link. Do you say that if i shift from 4.0.1 to 4.0.2 will fix the issue (of getting speaker icon in past conference)? --Manmohan Singh On Fri, Jul 30, 2010 at 8:16 PM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I did added the record option in user options as well. $Mod_Options = array(array(_(Announce), I), array(_(Record), r)); $User_Options = array(array(_(Announce), I), array(_(Listen Only), m), array(_(Wait for Leader), w), array(_(Record), r)); And the gre8 news is, it did worked this time. But it saved the recorded file in the following path: That is good to hear. /var/lib/asterisk/sounds/with the name as meetme-conf-rec-74438-1280463795.8.wav Than i tried to move the file to /var/lib/asterisk/sounds/conf-recordings/ just to see that it gives me a speaker icon when i click to past conferences. Unfortunately i couldnt see this speaker icon to hear this recorded conference .wav file. I am not surprised. By default MeetMe creates unique file names by appending pin-uniqueid, but uniqueid is not know until the conference starts, so the web interface does not know what to look for. Part of the changes to app_meetme included setting the realtime filename to use. I tried to download the .wav file into my windows machine and the filed played well.. like i mentioned in my earlier mail that following line i had added in lib/define.php, please correct me if i am wrong: define (RECORDING_PATH, /var/lib/asterisk/sounds/conf-recordings/); Do you think This recording path is taking the effect here? That setting effect where the WMM interface looks for recordings and not where Asterisk puts them. Looking back at your email history, I see you are on 4.0.1. After all of the suggestions, I remembered that I too found problems with recordings and addressed them in 4.0.2 That version adds a field to the database and stores the recording names in the database. I recommend using that version instead of 4.0.1. You can move your copy of lib/defines.php to the 4.0.2 install and keep your changes. Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] Asterisk Crashes - Segmentation Fault
Manmohan wrote: There was on very silly mistake and i missed to check that properly. Really apologize for that. Following change was done to get the conf-recording into the proper path: chown -R asterisk:asterisk /var/lib/asterisk/sounds/conf-recordings following is the output: [r...@linuxtest sounds]# ll total 6416 drwxrwxr-x 2 asterisk asterisk 4096 Jul 30 08:29 conf-recordings [r...@linuxtest sounds]# ll conf-recordings/ total 4060 -rw-r--r-- 1 asterisk asterisk 4150124 Jul 30 08:27 meetme-conf-rec-74438-1280463795.8.wav The only thing now is no speaker icon onto the webpage when i click to past conference link. The web interface cannot find the recording. The reason it cannot is that the name is wrong. By wrong, I mean it contains information that the database and program is not aware of (1280463795.8). To make this clear, if this conference was the 3rd one you ever scheduled on this system the correct file name would be- meetme-conf-rec-74438-3.wav using the format meetme-conf-rec-%PIN%-%BOOKID%.wav The database knows the pin and bookid, so it can construct the file name and test if it exists. Do you say that if i shift from 4.0.1 to 4.0.2 will fix the issue (of getting speaker icon in past conference)? I was not able to get the change into app_meetme to use the bookid in the filename, even though it has access to bookid. I gave up and now store the filename in the database, which app_meetme will use if it exists. Other that a handful of bug-fixes, this is the major difference between 4.0.1 and 4.0.2 Dan -- _ -- 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] Asterisk Crashes - Segmentation Fault
Manmohan wrote: Following is the output for core set verbose 5, also i am really not able to get on the admin pin thing? Do you mean, that with admin pin configured we cant use recording? You are actually running a version that has been fixed to support recording with pin-less or user pins. I should point out that the default settings in WMM is only to present the recording checkbox with the admin pin field. It is a fairly simple edit to add the recording checkbox to the user pin (and these options apply if no pin is set). Look in lib/defines.php for Mod_Options and User_Options to See how to add or remove MeetMe options from the GUI and database. Dan -- _ -- 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] Asterisk Crashes - Segmentation Fault
Hi Dan, I did added the record option in user options as well. $Mod_Options = array(array(_(Announce), I), array(_(Record), r)); $User_Options = array(array(_(Announce), I), array(_(Listen Only), m), array(_(Wait for Leader), w), array(_(Record), r)); And the gre8 news is, it did worked this time. But it saved the recorded file in the following path: /var/lib/asterisk/sounds/with the name as meetme-conf-rec-74438-1280463795.8.wav Than i tried to move the file to /var/lib/asterisk/sounds/conf-recordings/ just to see that it gives me a speaker icon when i click to past conferences. Unfortunately i couldnt see this speaker icon to hear this recorded conference .wav file. I tried to download the .wav file into my windows machine and the filed played well.. like i mentioned in my earlier mail that following line i had added in lib/define.php, please correct me if i am wrong: define (RECORDING_PATH, /var/lib/asterisk/sounds/conf-recordings/); Do you think This recording path is taking the effect here? --Manmohan Singh. On Fri, Jul 30, 2010 at 2:56 AM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: Following is the output for core set verbose 5, also i am really not able to get on the admin pin thing? Do you mean, that with admin pin configured we cant use recording? You are actually running a version that has been fixed to support recording with pin-less or user pins. I should point out that the default settings in WMM is only to present the recording checkbox with the admin pin field. It is a fairly simple edit to add the recording checkbox to the user pin (and these options apply if no pin is set). Look in lib/defines.php for Mod_Options and User_Options to See how to add or remove MeetMe options from the GUI and database. Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] Asterisk Crashes - Segmentation Fault
Manmohan wrote: I can see the path does exists but i cant see any recordings happening inn there. There are no files in it Following is the output: /var/lib/asterisk/sounds drwxrwxrwx 2 asterisk apache 4096 Jun 27 20:54 conf-recordings I hope m understandly this correctly but m sure m missing something here ;-) You did understand, and we have eliminated another of the possible issues. Are you assigning an admin pin to these conferences? There is a patch that allows recording pinless concenferences, but is has oddly not been merged yet. Try setting an admin pin. If that does not work, send the CLI output with core set verbose 5 as you dial in to the conference. Dan -- _ -- 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] Asterisk Crashes - Segmentation Fault
Hi Dan, Following is the output for core set verbose 5, also i am really not able to get on the admin pin thing? Do you mean, that with admin pin configured we cant use recording? LinuxTest*CLI core set verbose 5 Verbosity was 3 and is now 5 == Using SIP RTP CoS mark 5 -- Executing [...@callman_incoming:1] MeetMe(SIP/callman02-0002, ) in new stack -- SIP/callman02-0002 Playing 'conf-getconfno.ulaw' (language 'en') == Parsing '/etc/asterisk/meetme.conf': == Found -- Created MeetMe conference 1023 for conference '77972' -- SIP/callman02-0002 Playing 'conf-getpin.ulaw' (language 'en') Starting recording of MeetMe Conference 77972 into file .. -- SIP/callman02-0002 Playing 'vm-rec-name.ulaw' (language 'en') [Jul 29 09:16:19] WARNING[25049]: file.c:1160 ast_writefile: No such format '' -- SIP/callman02-0002 Playing 'beep.ulaw' (language 'en') -- x=0, open writing: /var/spool/asterisk/meetme/meetme-username-77972-1 format: sln, 0x9bea628 -- User ended message by pressing # -- SIP/callman02-0002 Playing 'auth-thankyou.ulaw' (language 'en') -- SIP/callman02-0002 Playing 'conf-onlyperson.ulaw' (language 'en') == Using SIP RTP CoS mark 5 -- Executing [...@callman_incoming:1] MeetMe(SIP/callman02-0003, ) in new stack -- SIP/callman02-0003 Playing 'conf-getconfno.ulaw' (language 'en') -- SIP/callman02-0003 Playing 'conf-getpin.ulaw' (language 'en') Starting recording of MeetMe Conference 77972 into file .. -- SIP/callman02-0003 Playing 'vm-rec-name.ulaw' (language 'en') -- SIP/callman02-0003 Playing 'beep.ulaw' (language 'en') -- x=0, open writing: /var/spool/asterisk/meetme/meetme-username-77972-2 format: sln, 0x9bea628 -- User ended message by pressing # -- SIP/callman02-0003 Playing 'auth-thankyou.ulaw' (language 'en') -- DAHDI/pseudo-736798397 Playing '/var/spool/asterisk/meetme/meetme-username-77972-2.slin' (language 'en') -- DAHDI/pseudo-736798397 Playing 'conf-hasjoin.ulaw' (language 'en') -- SIP/callman02-0003 Playing 'conf-placeintoconf.ulaw' (language 'en') == Spawn extension (callman_incoming, 493, 1) exited non-zero on 'SIP/callman02-0002' -- Executing [...@callman_incoming:1] Set(SIP/callman02-0002, CDR(bookId)=) in new stack -- Executing [...@callman_incoming:2] Set(SIP/callman02-0002, CDR(CIDnum)=281) in new stack -- Executing [...@callman_incoming:3] Set(SIP/callman02-0002, CDR(CIDname)=Manmohan Singh Jandu) in new stack -- DAHDI/pseudo-736798397 Playing '/var/spool/asterisk/meetme/meetme-username-77972-1.slin' (language 'en') -- SIP/callman02-0003 Playing 'conf-leaderhasleft.ulaw' (language 'en') -- DAHDI/pseudo-736798397 Playing 'conf-hasleft.ulaw' (language 'en') -- Hungup 'DAHDI/pseudo-923268627' -- Hungup 'DAHDI/pseudo-736798397' == Spawn extension (callman_incoming, 493, 1) exited non-zero on 'SIP/callman02-0003' -- Executing [...@callman_incoming:1] Set(SIP/callman02-0003, CDR(bookId)=) in new stack -- Executing [...@callman_incoming:2] Set(SIP/callman02-0003, CDR(CIDnum)=115) in new stack -- Executing [...@callman_incoming:3] Set(SIP/callman02-0003, CDR(CIDname)=cipc) in new stack On Thu, Jul 29, 2010 at 2:39 AM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I can see the path does exists but i cant see any recordings happening inn there. There are no files in it Following is the output: /var/lib/asterisk/sounds drwxrwxrwx 2 asterisk apache 4096 Jun 27 20:54 conf-recordings I hope m understandly this correctly but m sure m missing something here ;-) You did understand, and we have eliminated another of the possible issues. Are you assigning an admin pin to these conferences? There is a patch that allows recording pinless concenferences, but is has oddly not been merged yet. Try setting an admin pin. If that does not work, send the CLI output with core set verbose 5 as you dial in to the conference. Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] Asterisk Crashes - Segmentation Fault
Also following is what i am putting in lib/define.php define (RECORDING_PATH, /var/lib/asterisk/sounds/conf-recordings/); On Thu, Jul 29, 2010 at 9:20 AM, Manmohan Singh Jandu manmoha...@gmail.comwrote: Hi Dan, Following is the output for core set verbose 5, also i am really not able to get on the admin pin thing? Do you mean, that with admin pin configured we cant use recording? LinuxTest*CLI core set verbose 5 Verbosity was 3 and is now 5 == Using SIP RTP CoS mark 5 -- Executing [...@callman_incoming:1] MeetMe(SIP/callman02-0002, ) in new stack -- SIP/callman02-0002 Playing 'conf-getconfno.ulaw' (language 'en') == Parsing '/etc/asterisk/meetme.conf': == Found -- Created MeetMe conference 1023 for conference '77972' -- SIP/callman02-0002 Playing 'conf-getpin.ulaw' (language 'en') Starting recording of MeetMe Conference 77972 into file .. -- SIP/callman02-0002 Playing 'vm-rec-name.ulaw' (language 'en') [Jul 29 09:16:19] WARNING[25049]: file.c:1160 ast_writefile: No such format '' -- SIP/callman02-0002 Playing 'beep.ulaw' (language 'en') -- x=0, open writing: /var/spool/asterisk/meetme/meetme-username-77972-1 format: sln, 0x9bea628 -- User ended message by pressing # -- SIP/callman02-0002 Playing 'auth-thankyou.ulaw' (language 'en') -- SIP/callman02-0002 Playing 'conf-onlyperson.ulaw' (language 'en') == Using SIP RTP CoS mark 5 -- Executing [...@callman_incoming:1] MeetMe(SIP/callman02-0003, ) in new stack -- SIP/callman02-0003 Playing 'conf-getconfno.ulaw' (language 'en') -- SIP/callman02-0003 Playing 'conf-getpin.ulaw' (language 'en') Starting recording of MeetMe Conference 77972 into file .. -- SIP/callman02-0003 Playing 'vm-rec-name.ulaw' (language 'en') -- SIP/callman02-0003 Playing 'beep.ulaw' (language 'en') -- x=0, open writing: /var/spool/asterisk/meetme/meetme-username-77972-2 format: sln, 0x9bea628 -- User ended message by pressing # -- SIP/callman02-0003 Playing 'auth-thankyou.ulaw' (language 'en') -- DAHDI/pseudo-736798397 Playing '/var/spool/asterisk/meetme/meetme-username-77972-2.slin' (language 'en') -- DAHDI/pseudo-736798397 Playing 'conf-hasjoin.ulaw' (language 'en') -- SIP/callman02-0003 Playing 'conf-placeintoconf.ulaw' (language 'en') == Spawn extension (callman_incoming, 493, 1) exited non-zero on 'SIP/callman02-0002' -- Executing [...@callman_incoming:1] Set(SIP/callman02-0002, CDR(bookId)=) in new stack -- Executing [...@callman_incoming:2] Set(SIP/callman02-0002, CDR(CIDnum)=281) in new stack -- Executing [...@callman_incoming:3] Set(SIP/callman02-0002, CDR(CIDname)=Manmohan Singh Jandu) in new stack -- DAHDI/pseudo-736798397 Playing '/var/spool/asterisk/meetme/meetme-username-77972-1.slin' (language 'en') -- SIP/callman02-0003 Playing 'conf-leaderhasleft.ulaw' (language 'en') -- DAHDI/pseudo-736798397 Playing 'conf-hasleft.ulaw' (language 'en') -- Hungup 'DAHDI/pseudo-923268627' -- Hungup 'DAHDI/pseudo-736798397' == Spawn extension (callman_incoming, 493, 1) exited non-zero on 'SIP/callman02-0003' -- Executing [...@callman_incoming:1] Set(SIP/callman02-0003, CDR(bookId)=) in new stack -- Executing [...@callman_incoming:2] Set(SIP/callman02-0003, CDR(CIDnum)=115) in new stack -- Executing [...@callman_incoming:3] Set(SIP/callman02-0003, CDR(CIDname)=cipc) in new stack On Thu, Jul 29, 2010 at 2:39 AM, Dan Austin dan_aus...@phoenix.comwrote: Manmohan wrote: I can see the path does exists but i cant see any recordings happening inn there. There are no files in it Following is the output: /var/lib/asterisk/sounds drwxrwxrwx 2 asterisk apache 4096 Jun 27 20:54 conf-recordings I hope m understandly this correctly but m sure m missing something here ;-) You did understand, and we have eliminated another of the possible issues. Are you assigning an admin pin to these conferences? There is a patch that allows recording pinless concenferences, but is has oddly not been merged yet. Try setting an admin pin. If that does not work, send the CLI output with core set verbose 5 as you dial in to the conference. Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- Thanks Regards Manmohan Singh Jandu -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Hi Dan, I can see the path does exists but i cant see any recordings happening inn there. There are no files in it Following is the output: /var/lib/asterisk/sounds drwxrwxrwx 2 asterisk apache 4096 Jun 27 20:54 conf-recordings I hope m understandly this correctly but m sure m missing something here ;-) --Manmohan Singh. On Tue, Jul 27, 2010 at 3:03 AM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan Singh Jandu wrote: OK, now i added the column members in the table booking manually. and disabled selinux to have this working. Now i am struggling with recording option in webmeetme. Not sure on how to enable it, though m checking the checkbox while creating the conference. But where does this save and how to retrieve it? The location of the recordings is set in lib/defines.php as RECORDING_PATH, which defaults to /var/lib/asterisk/sounds/conf-recordings/ You can listen to the recordings after the conferences scheduled stop time by looking at the Past conferences page and clicking on the speaker icon next to the conference number. A couple of items to note- 1. You may have to check the path to ensure it exists and that the asterisk process can write to it. 2. Your web service accounts needs read permissions for that path 3. The speaker icon only displays if a recording exists. Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] Asterisk crashes to start if compiled with pbx_lua on latest updated CentOS
On Tue, Jul 27, 2010 at 12:45 AM, Faisal Hanif fai...@vopium.com wrote: Did any one got it solved? If yes how? Yes, read doc/backtrace.txt. It will explain how to generate an unoptimized backtrace, then uploaded it to the mailing list. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Asterisk Crashes - Segmentation Fault
Manmohan Singh Jandu wrote: Excellent! I finally got it working, it was ODBC drivers issue actually. Installed the proper compatible version and its working. I thought that might be the case. There are still few errors which i see on asterisk console: [Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:1032 require_odbc: Realtime table book...@meetme requires column 'members', but that column does not exist! WMM does not use that column. You can disable it by Setting logmembercount=no in meetme.conf Also when i try to click the conference to manage it realtime it gives me Error connection to the manager! Following are the database files which i used: /web-meetme/cbmysql/db-admin-user-create.txt /web-meetme/cbmysql/db-table-create-v6.txt /web-meetme/cbmysql/db-tables-v6.txt Am i missing something here now? The WMM web interface used the Asterisk manager interface to monitor and manage conferences. The readme file documents the required changes to manager.conf. Sorry for the delay responding, I was on vacation last week with no email access. Dan -- _ -- 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] Asterisk Crashes - Segmentation Fault
Manmohan Singh Jandu wrote: OK, now i added the column members in the table booking manually. and disabled selinux to have this working. Now i am struggling with recording option in webmeetme. Not sure on how to enable it, though m checking the checkbox while creating the conference. But where does this save and how to retrieve it? The location of the recordings is set in lib/defines.php as RECORDING_PATH, which defaults to /var/lib/asterisk/sounds/conf-recordings/ You can listen to the recordings after the conferences scheduled stop time by looking at the Past conferences page and clicking on the speaker icon next to the conference number. A couple of items to note- 1. You may have to check the path to ensure it exists and that the asterisk process can write to it. 2. Your web service accounts needs read permissions for that path 3. The speaker icon only displays if a recording exists. Dan -- _ -- 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] Asterisk crashes to start if compiled with pbx_lua on latest updated CentOS
Hi, I have tried number of time if we update any CentOS system (or use latest CentOS version) then compile asterisk 1.6.2 with pbx_lua support, asterisk will crash on starting and will give a core dump. Issue is easy to produce, Install latest CentOS on a system. Install LUA LUA Headers using YUM. Download and Compile latest release of asterisk 1.6.2. Try to start start asterisk in console mode. It will crash on LUA and will give a core dump Did any one got it solved? If yes how? Regards, Faisal Hanif -- _ -- 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] Asterisk Crashes - Segmentation Fault
Excellent! I finally got it working, it was ODBC drivers issue actually. Installed the proper compatible version and its working. There are still few errors which i see on asterisk console: [Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:1032 require_odbc: Realtime table book...@meetme requires column 'members', but that column does not exist! [Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:440 update_odbc: Key field 'members' does not exist in table 'book...@meetme'. Update will fail [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:616 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S22: [MySQL][ODBC 3.51 Driver][mysqld-5.0.77]Unknown column 'members' in 'field list' (80) [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:628 ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a reconnect... [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:723 ast_odbc_sanity_check: Connection is down attempting to reconnect... [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1427 odbc_obj_connect: Connecting meetme [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1455 odbc_obj_connect: res_odbc: Connected to meetme [meetme] [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:616 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S22: [MySQL][ODBC 3.51 Driver][mysqld-5.0.77]Unknown column 'members' in 'field list' (80) [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:628 ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a reconnect... [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:723 ast_odbc_sanity_check: Connection is down attempting to reconnect... [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1427 odbc_obj_connect: Connecting meetme [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1455 odbc_obj_connect: res_odbc: Connected to meetme [meetme] -- SIP/callman02-0005 Playing 'conf-onlyperson.ulaw' (language 'en') Also when i try to click the conference to manage it realtime it gives me Error connection to the manager! Following are the database files which i used: /web-meetme/cbmysql/db-admin-user-create.txt /web-meetme/cbmysql/db-table-create-v6.txt /web-meetme/cbmysql/db-tables-v6.txt Am i missing something here now? On Tue, Jul 13, 2010 at 8:43 PM, cov...@ccs.covici.com wrote: cov...@ccs.covici.com wrote: Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Good. There are other instruction packages, but since I wrote the README it is the one I am most familiar with. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk was in a separate Asterisk application (app_cbmysql). With version 4 of WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe application. The README in 4.0.1 lists the steps to setup RealTime (database) support for Asterisk and MeetMe. This narrows down the possible problems, since we do not need to consider app_cbmysql. Did you install Asterisk from a package with yum, or did you compile it yourself? Dan I am getting this error without webmeetme at all, after upgrading to svn-275706 from an earlier version 262801. Its a certain argument of meetme which I have not trafcked down yet which is causing this. OK, if the argument to meetme is conference number,TcMsrm it does not crash, but if it is conference number, cMs then it dies -- asterisk dies. Is this enough for someone to figure out? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] Asterisk Crashes - Segmentation Fault
OK, now i added the column members in the table booking manually. and disabled selinux to have this working. Now i am struggling with recording option in webmeetme. Not sure on how to enable it, though m checking the checkbox while creating the conference. But where does this save and how to retrieve it? On Mon, Jul 19, 2010 at 9:57 AM, Manmohan Singh Jandu manmoha...@gmail.comwrote: Excellent! I finally got it working, it was ODBC drivers issue actually. Installed the proper compatible version and its working. There are still few errors which i see on asterisk console: [Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:1032 require_odbc: Realtime table book...@meetme requires column 'members', but that column does not exist! [Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:440 update_odbc: Key field 'members' does not exist in table 'book...@meetme'. Update will fail [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:616 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S22: [MySQL][ODBC 3.51 Driver][mysqld-5.0.77]Unknown column 'members' in 'field list' (80) [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:628 ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a reconnect... [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:723 ast_odbc_sanity_check: Connection is down attempting to reconnect... [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1427 odbc_obj_connect: Connecting meetme [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1455 odbc_obj_connect: res_odbc: Connected to meetme [meetme] [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:616 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S22: [MySQL][ODBC 3.51 Driver][mysqld-5.0.77]Unknown column 'members' in 'field list' (80) [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:628 ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a reconnect... [Jul 19 13:58:51] WARNING[30213]: res_odbc.c:723 ast_odbc_sanity_check: Connection is down attempting to reconnect... [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1427 odbc_obj_connect: Connecting meetme [Jul 19 13:58:51] NOTICE[30213]: res_odbc.c:1455 odbc_obj_connect: res_odbc: Connected to meetme [meetme] -- SIP/callman02-0005 Playing 'conf-onlyperson.ulaw' (language 'en') Also when i try to click the conference to manage it realtime it gives me Error connection to the manager! Following are the database files which i used: /web-meetme/cbmysql/db-admin-user-create.txt /web-meetme/cbmysql/db-table-create-v6.txt /web-meetme/cbmysql/db-tables-v6.txt Am i missing something here now? On Tue, Jul 13, 2010 at 8:43 PM, cov...@ccs.covici.com wrote: cov...@ccs.covici.com wrote: Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Good. There are other instruction packages, but since I wrote the README it is the one I am most familiar with. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk was in a separate Asterisk application (app_cbmysql). With version 4 of WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe application. The README in 4.0.1 lists the steps to setup RealTime (database) support for Asterisk and MeetMe. This narrows down the possible problems, since we do not need to consider app_cbmysql. Did you install Asterisk from a package with yum, or did you compile it yourself? Dan I am getting this error without webmeetme at all, after upgrading to svn-275706 from an earlier version 262801. Its a certain argument of meetme which I have not trafcked down yet which is causing this. OK, if the argument to meetme is conference number,TcMsrm it does not crash, but if it is conference number, cMs then it dies -- asterisk dies. Is this enough for someone to figure out? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- Thanks Regards Manmohan Singh Jandu --
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Good. There are other instruction packages, but since I wrote the README it is the one I am most familiar with. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk was in a separate Asterisk application (app_cbmysql). With version 4 of WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe application. The README in 4.0.1 lists the steps to setup RealTime (database) support for Asterisk and MeetMe. This narrows down the possible problems, since we do not need to consider app_cbmysql. Did you install Asterisk from a package with yum, or did you compile it yourself? Dan I am getting this error without webmeetme at all, after upgrading to svn-275706 from an earlier version 262801. Its a certain argument of meetme which I have not trafcked down yet which is causing this. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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] Asterisk Crashes - Segmentation Fault
cov...@ccs.covici.com wrote: Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Good. There are other instruction packages, but since I wrote the README it is the one I am most familiar with. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk was in a separate Asterisk application (app_cbmysql). With version 4 of WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe application. The README in 4.0.1 lists the steps to setup RealTime (database) support for Asterisk and MeetMe. This narrows down the possible problems, since we do not need to consider app_cbmysql. Did you install Asterisk from a package with yum, or did you compile it yourself? Dan I am getting this error without webmeetme at all, after upgrading to svn-275706 from an earlier version 262801. Its a certain argument of meetme which I have not trafcked down yet which is causing this. OK, if the argument to meetme is conference number,TcMsrm it does not crash, but if it is conference number, cMs then it dies -- asterisk dies. Is this enough for someone to figure out? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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] Asterisk Crashes - Segmentation Fault
Manmohan wrote: Unfortunately m not able to get rid of the below mentioned errors. not sure on where i am missing now. On Sat, Jul 10, 2010 at 9:41 AM, Manmohan Singh Jandu manmoha...@gmail.com wrote: Ahh here is the catch i was still using app_cbmysql for this. now i had removed and just followed the README of 4.0 for WMM and m getting following on ,my asterisk console. Verbosity is at least 3 == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] MeetMe(SIP/492-, ) in new stack -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') == Parsing '/etc/asterisk/meetme.conf': == Found [Jul 10 13:42:15] NOTICE[16906]: res_odbc.c:1427 odbc_obj_connect: Connecting meetme [Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1452 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1273 ast_odbc_request_obj2: Failed to connect to meetme [Jul 10 13:42:15] ERROR[16906]: res_config_odbc.c:144 realtime_odbc: No database handle available with the name of 'meetme' (check res_odbc.conf) -- SIP/492- Playing 'conf-invalid.ulaw' (language 'en') -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') == Spawn extension (phones, 493, 1) exited non-zero on 'SIP/492-' (Initially i installed using yum, i was getting the same issue. Than i scrapped everything and installed it manually.) The good news is that you are making progress. Do you have the package unixODBC installed? The hint to that would have been if you created a new /etc/odbc.ini instead of editing a sample that the package would have installed. Dan -- _ -- 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] Asterisk Crashes - Segmentation Fault
Unfortunately m not able to get rid of the below mentioned errors. not sure on where i am missing now. On Sat, Jul 10, 2010 at 9:41 AM, Manmohan Singh Jandu manmoha...@gmail.comwrote: Ahh here is the catch i was still using app_cbmysql for this. now i had removed and just followed the README of 4.0 for WMM and m getting following on ,my asterisk console. Verbosity is at least 3 == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] MeetMe(SIP/492-, ) in new stack -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') == Parsing '/etc/asterisk/meetme.conf': == Found [Jul 10 13:42:15] NOTICE[16906]: res_odbc.c:1427 odbc_obj_connect: Connecting meetme [Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1452 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1273 ast_odbc_request_obj2: Failed to connect to meetme [Jul 10 13:42:15] ERROR[16906]: res_config_odbc.c:144 realtime_odbc: No database handle available with the name of 'meetme' (check res_odbc.conf) -- SIP/492- Playing 'conf-invalid.ulaw' (language 'en') -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') == Spawn extension (phones, 493, 1) exited non-zero on 'SIP/492-' (Initially i installed using yum, i was getting the same issue. Than i scrapped everything and installed it manually.) On Fri, Jul 9, 2010 at 8:39 PM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Good. There are other instruction packages, but since I wrote the README it is the one I am most familiar with. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk was in a separate Asterisk application (app_cbmysql). With version 4 of WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe application. The README in 4.0.1 lists the steps to setup RealTime (database) support for Asterisk and MeetMe. This narrows down the possible problems, since we do not need to consider app_cbmysql. Did you install Asterisk from a package with yum, or did you compile it yourself? Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] Asterisk Crashes - Segmentation Fault
Hi, My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. The GDB output is huge on, Following are my GDB errors. [r...@linuxtest tmp]# gdb asterisk core.LinuxTest-2010-07-07T21:13:15+0400 | more GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-23.el5_5.1) Copyright (C) 2009 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as i386-redhat-linux-gnu. For bug reporting instructions, please see: http://www.gnu.org/software/gdb/bugs/... Reading symbols from /usr/sbin/asterisk...done. warning: .dynamic section for /usr/lib/libidn.so.11 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations [New Thread 3212] SOME OF THE LINES IN the end of GDB Error: Reading symbols from /usr/lib/asterisk/modules/cdr_manager.so...done. Loaded symbols for /usr/lib/asterisk/modules/cdr_manager.so Reading symbols from /usr/lib/asterisk/modules/res_config_mysql.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/asterisk/modules/res_config_mysql.so Reading symbols from /usr/lib/asterisk/modules/chan_phone.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_phone.so Core was generated by `/usr/sbin/asterisk -f -vvvg -c'. Program terminated with signal 11, Segmentation fault. #0 0x01027d9d in mysql_fetch_row () from /usr/lib/mysql/libmysqlclient.so.15 --Manmohan Singh. On Thu, Jul 8, 2010 at 11:21 PM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I was looking for audio conferencing solution where i got Web-meetme. I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working fine. I tried using Meetme even meetme app is working perfectly fine. I installed Webmeetme 4.0 and integrated with my asterisk. When i try to dial the conference number it take me to an IVR wherein it asks for the conference number. The time i provide the conference number, asterisk crashes giving segmentation fault. I have been trying to google up and checked lot of forums but didnt get any solution for this yet. Which instructions did you follow for the integration? Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? Which exact version of WMM? Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] Asterisk Crashes - Segmentation Fault
Hi Install mysql 'n mysql-devel which includes /usr/lib/mysql/libmysqlclient.so.15 library. And also insert /usr/lib/mysql into /etc/ld.so.conf and then execute ldconfig command on terminal. -- Regards, Chandrakant Solanki On Fri, Jul 9, 2010 at 2:32 PM, Manmohan Singh Jandu manmoha...@gmail.comwrote: Hi, My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. The GDB output is huge on, Following are my GDB errors. [r...@linuxtest tmp]# gdb asterisk core.LinuxTest-2010-07-07T21:13:15+0400 | more GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-23.el5_5.1) Copyright (C) 2009 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as i386-redhat-linux-gnu. For bug reporting instructions, please see: http://www.gnu.org/software/gdb/bugs/... Reading symbols from /usr/sbin/asterisk...done. warning: .dynamic section for /usr/lib/libidn.so.11 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations [New Thread 3212] SOME OF THE LINES IN the end of GDB Error: Reading symbols from /usr/lib/asterisk/modules/cdr_manager.so...done. Loaded symbols for /usr/lib/asterisk/modules/cdr_manager.so Reading symbols from /usr/lib/asterisk/modules/res_config_mysql.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/asterisk/modules/res_config_mysql.so Reading symbols from /usr/lib/asterisk/modules/chan_phone.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_phone.so Core was generated by `/usr/sbin/asterisk -f -vvvg -c'. Program terminated with signal 11, Segmentation fault. #0 0x01027d9d in mysql_fetch_row () from /usr/lib/mysql/libmysqlclient.so.15 --Manmohan Singh. On Thu, Jul 8, 2010 at 11:21 PM, Dan Austin dan_aus...@phoenix.comwrote: Manmohan wrote: I was looking for audio conferencing solution where i got Web-meetme. I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working fine. I tried using Meetme even meetme app is working perfectly fine. I installed Webmeetme 4.0 and integrated with my asterisk. When i try to dial the conference number it take me to an IVR wherein it asks for the conference number. The time i provide the conference number, asterisk crashes giving segmentation fault. I have been trying to google up and checked lot of forums but didnt get any solution for this yet. Which instructions did you follow for the integration? Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? Which exact version of WMM? Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] Asterisk Crashes - Segmentation Fault
It still crashes and in gdb trace following is what its showing: --More-- warning: .dynamic section for /usr/lib/mysql/libmysqlclient.so.15 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations [New Thread 13310] LAST FEW LINES IN GDB TRACE: Reading symbols from /usr/lib/asterisk/modules/cdr_manager.so...done. Loaded symbols for /usr/lib/asterisk/modules/cdr_manager.so Reading symbols from /usr/lib/asterisk/modules/res_config_mysql.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/asterisk/modules/res_config_mysql.so Reading symbols from /usr/lib/asterisk/modules/chan_phone.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_phone.so Core was generated by `/usr/sbin/asterisk -f -vvvg -c'. Program terminated with signal 11, Segmentation fault. #0 0x003acd9d in mysql_fetch_row () from /usr/lib/mysql/libmysqlclient.so.15 --Manmohan Singh On Fri, Jul 9, 2010 at 2:36 PM, Manmohan Singh Jandu manmoha...@gmail.comwrote: Hi, Following is what i did. [r...@linuxtest ~]# yum install mysql* Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile * addons: centos.skknet.net * base: centos.skknet.net * extras: centos.skknet.net * updates: centos.skknet.net Setting up Install Process Package mysql-devel-5.0.77-4.el5_5.3.i386 already installed and latest version Package mysql-server-5.0.77-4.el5_5.3.i386 already installed and latest version Package mysql-5.0.77-4.el5_5.3.i386 already installed and latest version Package mysql-connector-odbc-3.51.26r1127-1.el5.i386 already installed and latest version Resolving Dependencies -- Running transaction check --- Package mysql-bench.i386 0:5.0.77-4.el5_5.3 set to be updated --- Package mysql-test.i386 0:5.0.77-4.el5_5.3 set to be updated -- Finished Dependency Resolution Dependencies Resolved PackageArchVersionRepository Size Installing: mysql-benchi3865.0.77-4.el5_5.3 updates 507 k mysql-test i3865.0.77-4.el5_5.3 updates 3.7 M Transaction Summary Install 2 Package(s) Upgrade 0 Package(s) Total download size: 4.2 M Is this ok [y/N]: y Downloading Packages: (1/2): mysql-bench-5.0.77-4.el5_5.3.i386.rpm | 507 kB 00:02 (2/2): mysql-test-5.0.77-4.el5_5.3.i386.rpm | 3.7 MB 00:11 Total 295 kB/s | 4.2 MB 00:14 Running rpm_check_debug Running Transaction Test Finished Transaction Test Transaction Test Succeeded Running Transaction Installing : mysql-bench 1/2 Installing : mysql-test 2/2 Installed: mysql-bench.i386 0:5.0.77-4.el5_5.3 mysql-test.i386 0:5.0.77-4.el5_5.3 Complete! [r...@linuxtest ~]# yum install mysql-devel* Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile * addons: centos.skknet.net * base: centos.skknet.net * extras: centosr4.centos.org * updates: centosg4.centos.org Setting up Install Process Package mysql-devel-5.0.77-4.el5_5.3.i386 already installed and latest version Nothing to do [r...@linuxtest ~]# cat /etc/ld.so.conf include ld.so.conf.d/*.conf [r...@linuxtest ~]# vi /etc/ld.so.conf [r...@linuxtest ~]# ldconfig [r...@linuxtest ~]# cat /etc/ld.so.conf include ld.so.conf.d/*.conf /usr/lib/mysql [r...@linuxtest ~]# ldconfig [r...@linuxtest ~]# Thanks Regards Manmohan Singh On Fri, Jul 9, 2010 at 2:19 PM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Hi Install mysql 'n mysql-devel which includes /usr/lib/mysql/libmysqlclient.so.15 library. And also insert /usr/lib/mysql into /etc/ld.so.conf and then execute ldconfig command on terminal. -- Regards, Chandrakant Solanki On Fri, Jul 9, 2010 at 2:32 PM, Manmohan Singh Jandu manmoha...@gmail.com wrote: Hi, My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. The GDB output is huge on, Following are my GDB errors. [r...@linuxtest tmp]# gdb asterisk core.LinuxTest-2010-07-07T21:13:15+0400 | more GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-23.el5_5.1) Copyright (C) 2009 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Hi, Following is what i did. [r...@linuxtest ~]# yum install mysql* Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile * addons: centos.skknet.net * base: centos.skknet.net * extras: centos.skknet.net * updates: centos.skknet.net Setting up Install Process Package mysql-devel-5.0.77-4.el5_5.3.i386 already installed and latest version Package mysql-server-5.0.77-4.el5_5.3.i386 already installed and latest version Package mysql-5.0.77-4.el5_5.3.i386 already installed and latest version Package mysql-connector-odbc-3.51.26r1127-1.el5.i386 already installed and latest version Resolving Dependencies -- Running transaction check --- Package mysql-bench.i386 0:5.0.77-4.el5_5.3 set to be updated --- Package mysql-test.i386 0:5.0.77-4.el5_5.3 set to be updated -- Finished Dependency Resolution Dependencies Resolved PackageArchVersionRepository Size Installing: mysql-benchi3865.0.77-4.el5_5.3 updates 507 k mysql-test i3865.0.77-4.el5_5.3 updates 3.7 M Transaction Summary Install 2 Package(s) Upgrade 0 Package(s) Total download size: 4.2 M Is this ok [y/N]: y Downloading Packages: (1/2): mysql-bench-5.0.77-4.el5_5.3.i386.rpm | 507 kB 00:02 (2/2): mysql-test-5.0.77-4.el5_5.3.i386.rpm | 3.7 MB 00:11 Total 295 kB/s | 4.2 MB 00:14 Running rpm_check_debug Running Transaction Test Finished Transaction Test Transaction Test Succeeded Running Transaction Installing : mysql-bench 1/2 Installing : mysql-test 2/2 Installed: mysql-bench.i386 0:5.0.77-4.el5_5.3 mysql-test.i386 0:5.0.77-4.el5_5.3 Complete! [r...@linuxtest ~]# yum install mysql-devel* Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile * addons: centos.skknet.net * base: centos.skknet.net * extras: centosr4.centos.org * updates: centosg4.centos.org Setting up Install Process Package mysql-devel-5.0.77-4.el5_5.3.i386 already installed and latest version Nothing to do [r...@linuxtest ~]# cat /etc/ld.so.conf include ld.so.conf.d/*.conf [r...@linuxtest ~]# vi /etc/ld.so.conf [r...@linuxtest ~]# ldconfig [r...@linuxtest ~]# cat /etc/ld.so.conf include ld.so.conf.d/*.conf /usr/lib/mysql [r...@linuxtest ~]# ldconfig [r...@linuxtest ~]# Thanks Regards Manmohan Singh On Fri, Jul 9, 2010 at 2:19 PM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Hi Install mysql 'n mysql-devel which includes /usr/lib/mysql/libmysqlclient.so.15 library. And also insert /usr/lib/mysql into /etc/ld.so.conf and then execute ldconfig command on terminal. -- Regards, Chandrakant Solanki On Fri, Jul 9, 2010 at 2:32 PM, Manmohan Singh Jandu manmoha...@gmail.com wrote: Hi, My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. The GDB output is huge on, Following are my GDB errors. [r...@linuxtest tmp]# gdb asterisk core.LinuxTest-2010-07-07T21:13:15+0400 | more GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-23.el5_5.1) Copyright (C) 2009 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as i386-redhat-linux-gnu. For bug reporting instructions, please see: http://www.gnu.org/software/gdb/bugs/... Reading symbols from /usr/sbin/asterisk...done. warning: .dynamic section for /usr/lib/libidn.so.11 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations [New Thread 3212] SOME OF THE LINES IN the end of GDB Error: Reading symbols from /usr/lib/asterisk/modules/cdr_manager.so...done. Loaded symbols for /usr/lib/asterisk/modules/cdr_manager.so Reading symbols from /usr/lib/asterisk/modules/res_config_mysql.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/asterisk/modules/res_config_mysql.so Reading symbols from /usr/lib/asterisk/modules/chan_phone.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_phone.so Core was generated by `/usr/sbin/asterisk -f -vvvg
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Manmohan wrote: My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Good. There are other instruction packages, but since I wrote the README it is the one I am most familiar with. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk was in a separate Asterisk application (app_cbmysql). With version 4 of WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe application. The README in 4.0.1 lists the steps to setup RealTime (database) support for Asterisk and MeetMe. This narrows down the possible problems, since we do not need to consider app_cbmysql. Did you install Asterisk from a package with yum, or did you compile it yourself? Dan -- _ -- 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] Asterisk Crashes - Segmentation Fault
Ahh here is the catch i was still using app_cbmysql for this. now i had removed and just followed the README of 4.0 for WMM and m getting following on ,my asterisk console. Verbosity is at least 3 == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] MeetMe(SIP/492-, ) in new stack -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') == Parsing '/etc/asterisk/meetme.conf': == Found [Jul 10 13:42:15] NOTICE[16906]: res_odbc.c:1427 odbc_obj_connect: Connecting meetme [Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1452 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1273 ast_odbc_request_obj2: Failed to connect to meetme [Jul 10 13:42:15] ERROR[16906]: res_config_odbc.c:144 realtime_odbc: No database handle available with the name of 'meetme' (check res_odbc.conf) -- SIP/492- Playing 'conf-invalid.ulaw' (language 'en') -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') == Spawn extension (phones, 493, 1) exited non-zero on 'SIP/492-' (Initially i installed using yum, i was getting the same issue. Than i scrapped everything and installed it manually.) On Fri, Jul 9, 2010 at 8:39 PM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Good. There are other instruction packages, but since I wrote the README it is the one I am most familiar with. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk was in a separate Asterisk application (app_cbmysql). With version 4 of WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe application. The README in 4.0.1 lists the steps to setup RealTime (database) support for Asterisk and MeetMe. This narrows down the possible problems, since we do not need to consider app_cbmysql. Did you install Asterisk from a package with yum, or did you compile it yourself? Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] Asterisk Crashes - Segmentation Fault
Hello Team, I was looking for audio conferencing solution where i got Web-meetme. I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working fine. I tried using Meetme even meetme app is working perfectly fine. I installed Webmeetme 4.0 and integrated with my asterisk. When i try to dial the conference number it take me to an IVR wherein it asks for the conference number. The time i provide the conference number, asterisk crashes giving segmentation fault. I have been trying to google up and checked lot of forums but didnt get any solution for this yet. Kernel version -- 2.6.18-194.3.1.el5PAE -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] Asterisk Crashes - Segmentation Fault
On Thu, Jul 8, 2010 at 12:21 PM, Manmohan Singh Jandu manmoha...@gmail.comwrote: Hello Team, I was looking for audio conferencing solution where i got Web-meetme. I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working fine. I tried using Meetme even meetme app is working perfectly fine. I installed Webmeetme 4.0 and integrated with my asterisk. When i try to dial the conference number it take me to an IVR wherein it asks for the conference number. The time i provide the conference number, asterisk crashes giving segmentation fault. I have been trying to google up and checked lot of forums but didnt get any solution for this yet. Kernel version -- 2.6.18-194.3.1.el5PAE -- Thanks Regards Manmohan Singh Jandu -- _ -- 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 Hi If you get Segmentation fault. One of core.$ file is created. Try to use # gdb asterisk core.$ and use bt command. And then paste error here. -- Regards, Chandrakant Solanki -- _ -- 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] Asterisk Crashes - Segmentation Fault
On Thu, Jul 8, 2010 at 2:51 AM, Manmohan Singh Jandu manmoha...@gmail.com wrote: crashes giving segmentation fault. Read doc/backtrace.txt on how to capture and generate a backtrace. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Asterisk Crashes - Segmentation Fault
Manmohan wrote: I was looking for audio conferencing solution where i got Web-meetme. I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working fine. I tried using Meetme even meetme app is working perfectly fine. I installed Webmeetme 4.0 and integrated with my asterisk. When i try to dial the conference number it take me to an IVR wherein it asks for the conference number. The time i provide the conference number, asterisk crashes giving segmentation fault. I have been trying to google up and checked lot of forums but didnt get any solution for this yet. Which instructions did you follow for the integration? Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? Which exact version of WMM? Dan -- _ -- 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] Asterisk crashes : Failed to start PBX
Hi Neo, have you checked your log files? It sometimes happened to me that Asterisk crashed without a reason. I discovered my logrotate didn't make its dirty work so I had huge log files. I lowered Asterisk log level and forced logrotate to work and now I have no more crashes. Hope it may help. :) Giorgio. Neo Anderson wrote: Hello, I am using Asterisk 1.4.24.1 version in production. OS is Centos 5.3 64 bit RAM is 8 GB. I am facing crash in asterisk approx each 12 hour. When it crashes I see below lines in asterisk logs. [Nov 18 06:47:23] WARNING[8730] pbx.c: Failed to create new channel thread [Nov 18 06:47:23] WARNING[8730] chan_sip.c: Failed to start PBX :( I debugged asterisk source code in details I found that it happens because it can not allocate memory to create thread. Another thing is, when I check coredump using gdb, it's not showing any debug symbols. Would you please let me know how to prevent or resolve this? Thanks in advance!! -- Regards, voipexpert ___ -- 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 ___ -- 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] Asterisk crashes : Failed to start PBX
Hi, I have already setup to rotate logs hourly Debug level is 3. Is there any other possibility of crash? Thanks in advance!! -- Regards, Voipexpert From: Giorgio Incantalupo gincantal...@fgasoftware.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Fri, November 20, 2009 10:06:06 PM Subject: Re: [asterisk-users] Asterisk crashes : Failed to start PBX Hi Neo, have you checked your log files? It sometimes happened to me that Asterisk crashed without a reason. I discovered my logrotate didn't make its dirty work so I had huge log files. I lowered Asterisk log level and forced logrotate to work and now I have no more crashes. Hope it may help. :) Giorgio. Neo Anderson wrote: Hello, I am using Asterisk 1.4.24.1 version in production. OS is Centos 5.3 64 bit RAM is 8 GB. I am facing crash in asterisk approx each 12 hour. When it crashes I see below lines in asterisk logs. [Nov 18 06:47:23] WARNING[8730] pbx.c: Failed to create new channel thread [Nov 18 06:47:23] WARNING[8730] chan_sip.c: Failed to start PBX :( I debugged asterisk source code in details I found that it happens because it can not allocate memory to create thread. Another thing is, when I check coredump using gdb, it's not showing any debug symbols. Would you please let me know how to prevent or resolve this? Thanks in advance!! -- Regards, voipexpert ___ -- 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 ___ -- 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 ___ -- 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] Asterisk crashes : Failed to start PBX
Hello, I am using Asterisk 1.4.24.1 version in production. OS is Centos 5.3 64 bit RAM is 8 GB. I am facing crash in asterisk approx each 12 hour. When it crashes I see below linesin asterisk logs. [Nov 18 06:47:23] WARNING[8730] pbx.c: Failed to create new channel thread [Nov 18 06:47:23] WARNING[8730] chan_sip.c: Failed to start PBX :( I debugged asterisk source code in details I found that it happens because it can not allocate memory to create thread. Another thing is, when I check coredump using gdb, it's not showing any debug symbols. Would you please let me know how to prevent or resolve this? Thanks in advance!! -- Regards, voipexpert ___ -- 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] asterisk crashes when calling gtalk user
Hi all, I'm using Asterisk 1.4.26.2. Every time I call a gtalk user, Asterisk crashes: -- Executing [6...@inbound:1] NoOp(SIP/8-084894d8, jabbertest) in new stack -- Executing [6...@inbound:2] Dial(SIP/8-084894d8, Gtalk/asterisk/pippopi...@gmail.com) in new stack Segmentation fault Is anybody experiencing the same? Found any workaround? Thank you. Giorgio Incantalupo ___ -- 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] asterisk crashes!!!
Hi, I got ast. 1.6.0.10 working for a few weeks without a problem. A few mins ago..I got the following msgs on ast-cli and asterisk service crashed. I coudlnt find anything that might cause this problem. Any ideas?? [Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein: Invalid GSM data (1) [Aug 7 13:54:34] WARNING[10102]: translate.c:204 framein: gsmtolin did not update samples 0 [Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein: Invalid GSM data (1) [Aug 7 13:54:34] WARNING[10102]: translate.c:204 framein: gsmtolin did not update samples 0 [Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein: Invalid GSM data (1) [Aug 7 13:54:34] WARNING[10102]: translate.c:204 framein: gsmtolin did not update samples 0 [Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein: Invalid GSM data (1) [Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein: Invalid GSM data (1) [Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein: Invalid GSM data (1) [Aug 7 13:54:40] WARNING[9517]: file.c:718 ast_readaudio_callback: Failed to write frame asterisk1*CLI Disconnected from Asterisk server ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk crashes
Hello, I have very annoying problem with asterisk 1.4.4: Every evening when I have peak load asterisk crashes, peak load is only over 20-30 sip-to-h323 simultaneous calls. Nothing special in logs after crash. Load average never was higher than 0.3, asterisk never uses more than 12% CPU (according to top). Tried SVN versions - same result. Both h323 and sip peers has only one codec allowed - g729 - so no conversion. There is no conferences, call recordings or something like this - very simple setup. Software config: Linux Slackware 11.0 Kernel 2.6.21.1 Asterisk 1.4.4 (native h323 channel from asterisk tarball) Libpri-1.4.0 Zaptel-1.4.2.1 (using ztdummy for internal sync, no zaptel hardware) pwlib_v1_10_0 openh323_v1_18_0 Hardware config: Intel SE7210TP1 motherboard P4 3GHz HT 1Mb cache CPU 1Gb RAM (dual channel, two same DIMMs from intel recommended list) 80Gb SATA HDD No zaptel hardware or even any PCI cards There isn't overheating and voltage problems with a hardware (controlling over IPMI), this hardware (with another HDD and software versions) worked fine about year with asterisk restarts manually only for a version upgrade. Could somebody point me the way to debug this problem? Thank you! -- Sincerely, Elman Efendiyev ___ --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] Asterisk crashes at startup
Hi List, Yesterday night after a power off due to a faulty UPS my asterisk doesn't want to start anymore. Here is what I get on the CLI: Asterisk Ready. *CLI Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk uncleanly ending (0). I use 1.2.7 I think on a debian sarge and cdr_pgsql too. Any ideas? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] Asterisk crashes at startup
You may have file damage. Run the file repair. Bob Rawlinson Jean-Michel Hiver wrote: Hi List, Yesterday night after a power off due to a faulty UPS my asterisk doesn't want to start anymore. Here is what I get on the CLI: Asterisk Ready. *CLI Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk uncleanly ending (0). I use 1.2.7 I think on a debian sarge and cdr_pgsql too. Any ideas? Cheers, Jean-Michel. ___ --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] Asterisk crashes at startup
run asterisk with asterisk -c and see if it gives anymore information. You can also get it to produce a core dump and see if it gives you anymore information. brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Wednesday, May 31, 2006 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk crashes at startup Hi List, Yesterday night after a power off due to a faulty UPS my asterisk doesn't want to start anymore. Here is what I get on the CLI: Asterisk Ready. *CLI Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk uncleanly ending (0). I use 1.2.7 I think on a debian sarge and cdr_pgsql too. Any ideas? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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 This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --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] Asterisk crashes at startup
look the list of hardware! http://www.voip-info.org/wiki/view/How+To+Debug+and+Troubleshoot+VOIP - Original Message - From: Jean-Michel Hiver [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 31, 2006 11:09 AM Subject: [Asterisk-Users] Asterisk crashes at startup Hi List, Yesterday night after a power off due to a faulty UPS my asterisk doesn't want to start anymore. Here is what I get on the CLI: Asterisk Ready. *CLI Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk uncleanly ending (0). I use 1.2.7 I think on a debian sarge and cdr_pgsql too. Any ideas? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.0/353 - Release Date: 31/05/2006 __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --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] Asterisk crashes at startup
Jean-Michel Hiver wrote: Hi List, Yesterday night after a power off due to a faulty UPS my asterisk doesn't want to start anymore. Here is what I get on the CLI: Asterisk Ready. *CLI Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk uncleanly ending (0). I use 1.2.7 I think on a debian sarge and cdr_pgsql too. Any ideas? Maybe the /var/run/asterisk.ctl file still exists and points nowhere? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] Asterisk crashes at startup
Robert Rawlinson a écrit : You may have file damage. Run the file repair. Rob, thanks for your response. Which tool would you use to do that? -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] Asterisk crashes at startup
Brian C. Fertig a écrit : run asterisk with asterisk -c and see if it gives anymore information. You can also get it to produce a core dump and see if it gives you anymore information. That's pretty much what I did, and it says: [func_uri.so] = (URI encode/decode functions) == Registered custom function URIDECODE == Registered custom function URIENCODE == Manager registered action DBGet == Manager registered action DBPut == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk cleanly ending (0). Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] Asterisk Crashes after update
Thanks Brian: Sorry for the late reply. That was helpful. Problem solved. Scott At 10:43 PM 7/10/2005, you wrote: You might want to recompile the res_config_mysql or configure res_config_odbc which works via myodbc and is just as good! /b --- Anakin: You're either with me, or you're my enemy. Obi-Wan: Only a Sith could be an absolutist. On Jul 7, 2005, at 2:46 PM, [EMAIL PROTECTED] wrote: After doing an update from SUSE 9.2 to 9.3 and Checking out the latest from CVS, Asterisk crashes on startup with an apparent MySQL (res_config_register) error: # asterisk -vvvgc asterisk_startup_error1.log asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_mysql.so: un defined symbol: ast_cust_config_register The log is shown below. I've seen the posts from 1/25/05 and several more recent ones regarding this same issue or a similar one with the ast_cust_config_register being undefined, however reverting to that build of 1/24/05 does not solve the problem in my case. Is there another issue with mySQL that may cause this problem? I'm using SUSE 9.3 on an Athlon 64 with 64 bit release 2.6 of Linux. I've made sure that all the ODBC and MySQL modules for SUSE 9.3 are installed. I'm a rank noob with * and would appreciate any help. Thanks!!! Log Pasted below for more info: [0;37;40m [1;30;40m == [0;37;40mParsing '/etc/asterisk/asterisk.conf': Found [1;30;40m == [0;37;40mParsing '/etc/asterisk/extconfig.conf': Found [1;30;40m == [0;37;40mParsing '/etc/asterisk/asterisk.conf': Found Asterisk CVS-HEAD, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] == === [1;30;40m == [0;37;40mParsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk Dynamic Loader loading preload modules: [1;30;40m == [0;37;40mParsing '/etc/asterisk/modules.conf': Found [1;30;40m == [0;37;40mManager registered action Ping [1;30;40m == [0;37;40mManager registered action Events [1;30;40m == [0;37;40mManager registered action Logoff [1;30;40m == [0;37;40mManager registered action Hangup [1;30;40m == [0;37;40mManager registered action Status [1;30;40m == [0;37;40mManager registered action Setvar [1;30;40m == [0;37;40mManager registered action Getvar [1;30;40m == [0;37;40mManager registered action Redirect [1;30;40m == [0;37;40mManager registered action Originate [1;30;40m == [0;37;40mManager registered action Command [1;30;40m == [0;37;40mManager registered action ExtensionState [1;30;40m == [0;37;40mManager registered action AbsoluteTimeout [1;30;40m == [0;37;40mManager registered action MailboxStatus [1;30;40m == [0;37;40mManager registered action MailboxCount [1;30;40m == [0;37;40mManager registered action ListCommands [1;30;40m == [0;37;40mParsing '/etc/asterisk/manager.conf': Found [1;30;40m == [0;37;40mParsing '/etc/asterisk/cdr.conf': Not found (No such file or directory) Jul 6 21:32:24 [1;33;40mNOTICE[0;37;40m[8492]: [1;37;40mcdr.c[0;37;40m:[1;37;40m1162[0;37;40m [1;37;40mdo_reload[0;37;40m: CDR simple logging enabled. [1;30;40m == [0;37;40mParsing '/etc/asterisk/rtp.conf': Found [1;30;40m == [0;37;40mRTP Allocating from port range 1 - 2 Asterisk PBX Core Initializing Registering builtin applications: [1;30;40m [0;37;40m[AbsoluteTimeout] [1;30;40m == [0;37;40mRegistered application '[1;36;40mAbsoluteTimeout[0;37;40m' [1;30;40m [0;37;40m[Answer] [1;30;40m == [0;37;40mRegistered application '[1;36;40mAnswer[0;37;40m' [1;30;40m [0;37;40m[BackGround] [1;30;40m == [0;37;40mRegistered application '[1;36;40mBackGround[0;37;40m' [1;30;40m [0;37;40m[Busy] [1;30;40m == [0;37;40mRegistered application '[1;36;40mBusy[0;37;40m' [1;30;40m [0;37;40m[Congestion] [1;30;40m == [0;37;40mRegistered application '[1;36;40mCongestion[0;37;40m' [1;30;40m [0;37;40m[DigitTimeout] [1;30;40m == [0;37;40mRegistered application '[1;36;40mDigitTimeout[0;37;40m' [1;30;40m [0;37;40m[Goto] [1;30;40m == [0;37;40mRegistered application '[1;36;40mGoto[0;37;40m' [1;30;40m [0;37;40m[GotoIf] [1;30;40m == [0;37;40mRegistered application '[1;36;40mGotoIf[0;37;40m' [1;30;40m [0;37;40m[GotoIfTime] [1;30;40m == [0;37;40mRegistered application '[1;36;40mGotoIfTime[0;37;40m' [1;30;40m [0;37;40m[ExecIfTime] [1;30;40m == [0;37;40mRegistered application '[1;36;40mExecIfTime[0;37;40m' [1;30;40m [0;37;40m[Hangup] [1;30;40m == [0;37;40mRegistered application '[1;36;40mHangup[0;37;40m' [1;30;40m [0;37;40m[NoOp] [1;30;40m == [0;37;40mRegistered application '[1;36;40mNoOp[0;37;40m' [1;30;40m [0;37;40m[Prefix] [1;30;40m == [0;37;40mRegistered application '[1;36;40mPrefix[0;37;40m' [1;30;40m [0;37;40m[Progress] [1;30;40m == [0;37;40mRegistered application '[1;36;40mProgress[0;37;40m' [1;30;40m [0;37;40m[ResetCDR] [1;30;40m == [0;37;40mRegistered application '[1;36;40mResetCDR[0;37;40m' [1;30;40m [0;37;40m[ResponseTimeout] [1;30;40m == [0;37;40mRegistered application
Re: [Asterisk-Users] Asterisk Crashes after update
You might want to recompile the res_config_mysql or configure res_config_odbc which works via myodbc and is just as good! /b --- Anakin: “You’re either with me, or you’re my enemy.” Obi-Wan: “Only a Sith could be an absolutist.” On Jul 7, 2005, at 2:46 PM, [EMAIL PROTECTED] wrote: After doing an update from SUSE 9.2 to 9.3 and Checking out the latest from CVS, Asterisk crashes on startup with an apparent MySQL (res_config_register) error: # asterisk -vvvgc asterisk_startup_error1.log asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_mysql.so: un defined symbol: ast_cust_config_register The log is shown below. I've seen the posts from 1/25/05 and several more recent ones regarding this same issue or a similar one with the ast_cust_config_register being undefined, however reverting to that build of 1/24/05 does not solve the problem in my case. Is there another issue with mySQL that may cause this problem? I'm using SUSE 9.3 on an Athlon 64 with 64 bit release 2.6 of Linux. I've made sure that all the ODBC and MySQL modules for SUSE 9.3 are installed. I'm a rank noob with * and would appreciate any help. Thanks!!! Log Pasted below for more info: [0;37;40m [1;30;40m == [0;37;40mParsing '/etc/asterisk/asterisk.conf': Found [1;30;40m == [0;37;40mParsing '/etc/asterisk/extconfig.conf': Found [1;30;40m == [0;37;40mParsing '/etc/asterisk/asterisk.conf': Found Asterisk CVS-HEAD, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] == === [1;30;40m == [0;37;40mParsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk Dynamic Loader loading preload modules: [1;30;40m == [0;37;40mParsing '/etc/asterisk/modules.conf': Found [1;30;40m == [0;37;40mManager registered action Ping [1;30;40m == [0;37;40mManager registered action Events [1;30;40m == [0;37;40mManager registered action Logoff [1;30;40m == [0;37;40mManager registered action Hangup [1;30;40m == [0;37;40mManager registered action Status [1;30;40m == [0;37;40mManager registered action Setvar [1;30;40m == [0;37;40mManager registered action Getvar [1;30;40m == [0;37;40mManager registered action Redirect [1;30;40m == [0;37;40mManager registered action Originate [1;30;40m == [0;37;40mManager registered action Command [1;30;40m == [0;37;40mManager registered action ExtensionState [1;30;40m == [0;37;40mManager registered action AbsoluteTimeout [1;30;40m == [0;37;40mManager registered action MailboxStatus [1;30;40m == [0;37;40mManager registered action MailboxCount [1;30;40m == [0;37;40mManager registered action ListCommands [1;30;40m == [0;37;40mParsing '/etc/asterisk/manager.conf': Found [1;30;40m == [0;37;40mParsing '/etc/asterisk/cdr.conf': Not found (No such file or directory) Jul 6 21:32:24 [1;33;40mNOTICE[0;37;40m[8492]: [1;37;40mcdr.c[0;37;40m:[1;37;40m1162[0;37;40m [1;37;40mdo_reload[0;37;40m: CDR simple logging enabled. [1;30;40m == [0;37;40mParsing '/etc/asterisk/rtp.conf': Found [1;30;40m == [0;37;40mRTP Allocating from port range 1 - 2 Asterisk PBX Core Initializing Registering builtin applications: [1;30;40m [0;37;40m[AbsoluteTimeout] [1;30;40m == [0;37;40mRegistered application '[1;36;40mAbsoluteTimeout[0;37;40m' [1;30;40m [0;37;40m[Answer] [1;30;40m == [0;37;40mRegistered application '[1;36;40mAnswer[0;37;40m' [1;30;40m [0;37;40m[BackGround] [1;30;40m == [0;37;40mRegistered application '[1;36;40mBackGround[0;37;40m' [1;30;40m [0;37;40m[Busy] [1;30;40m == [0;37;40mRegistered application '[1;36;40mBusy[0;37;40m' [1;30;40m [0;37;40m[Congestion] [1;30;40m == [0;37;40mRegistered application '[1;36;40mCongestion[0;37;40m' [1;30;40m [0;37;40m[DigitTimeout] [1;30;40m == [0;37;40mRegistered application '[1;36;40mDigitTimeout[0;37;40m' [1;30;40m [0;37;40m[Goto] [1;30;40m == [0;37;40mRegistered application '[1;36;40mGoto[0;37;40m' [1;30;40m [0;37;40m[GotoIf] [1;30;40m == [0;37;40mRegistered application '[1;36;40mGotoIf[0;37;40m' [1;30;40m [0;37;40m[GotoIfTime] [1;30;40m == [0;37;40mRegistered application '[1;36;40mGotoIfTime[0;37;40m' [1;30;40m [0;37;40m[ExecIfTime] [1;30;40m == [0;37;40mRegistered application '[1;36;40mExecIfTime[0;37;40m' [1;30;40m [0;37;40m[Hangup] [1;30;40m == [0;37;40mRegistered application '[1;36;40mHangup[0;37;40m' [1;30;40m [0;37;40m[NoOp] [1;30;40m == [0;37;40mRegistered application '[1;36;40mNoOp[0;37;40m' [1;30;40m [0;37;40m[Prefix] [1;30;40m == [0;37;40mRegistered application '[1;36;40mPrefix[0;37;40m' [1;30;40m [0;37;40m[Progress] [1;30;40m == [0;37;40mRegistered application '[1;36;40mProgress[0;37;40m' [1;30;40m [0;37;40m[ResetCDR] [1;30;40m == [0;37;40mRegistered application '[1;36;40mResetCDR[0;37;40m' [1;30;40m [0;37;40m[ResponseTimeout]
[Asterisk-Users] Asterisk Crashes after update
After doing an update from SUSE 9.2 to 9.3 and Checking out the latest from CVS, Asterisk crashes on startup with an apparent MySQL (res_config_register) error: # asterisk -vvvgc asterisk_startup_error1.log asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_mysql.so: un defined symbol: ast_cust_config_register The log is shown below. I've seen the posts from 1/25/05 and several more recent ones regarding this same issue or a similar one with the ast_cust_config_register being undefined, however reverting to that build of 1/24/05 does not solve the problem in my case. Is there another issue with mySQL that may cause this problem? I'm using SUSE 9.3 on an Athlon 64 with 64 bit release 2.6 of Linux. I've made sure that all the ODBC and MySQL modules for SUSE 9.3 are installed. I'm a rank noob with * and would appreciate any help. Thanks!!! Log Pasted below for more info: [0;37;40m [1;30;40m == [0;37;40mParsing '/etc/asterisk/asterisk.conf': Found [1;30;40m == [0;37;40mParsing '/etc/asterisk/extconfig.conf': Found [1;30;40m == [0;37;40mParsing '/etc/asterisk/asterisk.conf': Found Asterisk CVS-HEAD, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] = [1;30;40m == [0;37;40mParsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk Dynamic Loader loading preload modules: [1;30;40m == [0;37;40mParsing '/etc/asterisk/modules.conf': Found [1;30;40m == [0;37;40mManager registered action Ping [1;30;40m == [0;37;40mManager registered action Events [1;30;40m == [0;37;40mManager registered action Logoff [1;30;40m == [0;37;40mManager registered action Hangup [1;30;40m == [0;37;40mManager registered action Status [1;30;40m == [0;37;40mManager registered action Setvar [1;30;40m == [0;37;40mManager registered action Getvar [1;30;40m == [0;37;40mManager registered action Redirect [1;30;40m == [0;37;40mManager registered action Originate [1;30;40m == [0;37;40mManager registered action Command [1;30;40m == [0;37;40mManager registered action ExtensionState [1;30;40m == [0;37;40mManager registered action AbsoluteTimeout [1;30;40m == [0;37;40mManager registered action MailboxStatus [1;30;40m == [0;37;40mManager registered action MailboxCount [1;30;40m == [0;37;40mManager registered action ListCommands [1;30;40m == [0;37;40mParsing '/etc/asterisk/manager.conf': Found [1;30;40m == [0;37;40mParsing '/etc/asterisk/cdr.conf': Not found (No such file or directory) Jul 6 21:32:24 [1;33;40mNOTICE[0;37;40m[8492]: [1;37;40mcdr.c[0;37;40m:[1;37;40m1162[0;37;40m [1;37;40mdo_reload[0;37;40m: CDR simple logging enabled. [1;30;40m == [0;37;40mParsing '/etc/asterisk/rtp.conf': Found [1;30;40m == [0;37;40mRTP Allocating from port range 1 - 2 Asterisk PBX Core Initializing Registering builtin applications: [1;30;40m [0;37;40m[AbsoluteTimeout] [1;30;40m == [0;37;40mRegistered application '[1;36;40mAbsoluteTimeout[0;37;40m' [1;30;40m [0;37;40m[Answer] [1;30;40m == [0;37;40mRegistered application '[1;36;40mAnswer[0;37;40m' [1;30;40m [0;37;40m[BackGround] [1;30;40m == [0;37;40mRegistered application '[1;36;40mBackGround[0;37;40m' [1;30;40m [0;37;40m[Busy] [1;30;40m == [0;37;40mRegistered application '[1;36;40mBusy[0;37;40m' [1;30;40m [0;37;40m[Congestion] [1;30;40m == [0;37;40mRegistered application '[1;36;40mCongestion[0;37;40m' [1;30;40m [0;37;40m[DigitTimeout] [1;30;40m == [0;37;40mRegistered application '[1;36;40mDigitTimeout[0;37;40m' [1;30;40m [0;37;40m[Goto] [1;30;40m == [0;37;40mRegistered application '[1;36;40mGoto[0;37;40m' [1;30;40m [0;37;40m[GotoIf] [1;30;40m == [0;37;40mRegistered application '[1;36;40mGotoIf[0;37;40m' [1;30;40m [0;37;40m[GotoIfTime] [1;30;40m == [0;37;40mRegistered application '[1;36;40mGotoIfTime[0;37;40m' [1;30;40m [0;37;40m[ExecIfTime] [1;30;40m == [0;37;40mRegistered application '[1;36;40mExecIfTime[0;37;40m' [1;30;40m [0;37;40m[Hangup] [1;30;40m == [0;37;40mRegistered application '[1;36;40mHangup[0;37;40m' [1;30;40m [0;37;40m[NoOp] [1;30;40m == [0;37;40mRegistered application '[1;36;40mNoOp[0;37;40m' [1;30;40m [0;37;40m[Prefix] [1;30;40m == [0;37;40mRegistered application '[1;36;40mPrefix[0;37;40m' [1;30;40m [0;37;40m[Progress] [1;30;40m == [0;37;40mRegistered application '[1;36;40mProgress[0;37;40m' [1;30;40m [0;37;40m[ResetCDR] [1;30;40m == [0;37;40mRegistered application '[1;36;40mResetCDR[0;37;40m' [1;30;40m [0;37;40m[ResponseTimeout] [1;30;40m == [0;37;40mRegistered application '[1;36;40mResponseTimeout[0;37;40m' [1;30;40m
[Asterisk-Users] asterisk crashes
hi ! i have the following in my extensions.conf exten = 2000,1,Wait(60) exten = 2000,2,Hangup When i dial '2000' from my phone, I see 'Wait' being called. After 60 secs, I also se 'Hangup' being called. If I hangup the phone line before 60 secs are over ('Wait' command is probably interrupted in this case), asterisk crashes with segmentation fault. Due to this problem, my 'campon' feature causes asterisk to crash often. does anyone have an idea as to what this problem might be ? tulika _ Millions of marriage proposals. http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?74 Find your match on BharatMatrimony.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk crashes with sipp
with asterisk running, if i call sipp ip address -s 9111 -d 6 -r 20 -t un -sn uac -m 60 all the calls get set up, and after a minute when asterisk receives the 1st BYE from uac, it responds with 200 OK and then crashes. If i restart asterisk, all the calls get terminated properly. in the extensions, i have [default] exten = 9111222,1,Answer exten = 9111222,3,Wait(600) exten = 9111222,4,Hangup please help as i am unable to continue with any load tests ! tulika _ Kareena or Rani? Saif or SRK? http://server1.msn.co.in/sp05/iifa/ Rock your vote now at IIFA. ___ 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] Asterisk crashes
Can someone please help me. I am currently HEAD as of about 5 days ago (stable was giving me all sort of problems, upgraded per other users suggestions) on an Intel mainboard using a mix of Cisco 7960/40 SIP and 7910 SCCP. Can someone please explain what the following means? When this happens, I am about 1 minute from Asterisk going downhill. All of the SCCP phones quit, while the SIP phones can do calling to some degree. I get kicked out of any consoles and can't reconnect without restarting asterisk. Mark May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 DEBUG[28400] channel.c: Avoiding deadlock for 'SCCP/118-001a' May 7 01:03:13 WARNING[28400] channel.c: Avoided deadlock for 'SCCP/118-001a', 10 retries! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk crashes on special Transfer with MGCP/ATA 186
Thomas Dingermann wrote: Hi all, i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco ATA-186 3.1.1 atamgcp We are used to make an special ;) blind transfer like (Flash)Number(Hangup before anyone answers or ring). Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp If one waits until the last one rings, then hangup, everything is fine. If one waits until the last one answers, then hangup, everything is fine, too. Any hints? mgcp debug on: -- Executing AGI(Zap/7-1, nuller.agi) in new stack -- Launched AGI Script /home/kpj/pbx/var/lib/asterisk/agi-bin/nuller.agi -- Accepting call from '01635571857' to '8551' on channel 0/1, span 3 -- AGI Script nuller.agi completed, returning 0 -- Executing Dial(Zap/7-1, MGCP/aaln/[EMAIL PROTECTED]||) in new stack -- MGCP mgcp_request(aaln/[EMAIL PROTECTED]) -- MGCP cw: 0, dnd: 0, so: 0, sno: 0 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Called aaln/[EMAIL PROTECTED] -- MGCP/aaln/[EMAIL PROTECTED] is ringing -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP/aaln/[EMAIL PROTECTED] answered Zap/7-1 gw-bzo*CLI mgcp debug on Usage: mgcp debug Enables dumping of MGCP packets for debugging purposes -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf' -- Swapping 1 for 0 on aaln/[EMAIL PROTECTED] -- MGCP Muting 1 on aaln/[EMAIL PROTECTED] -- Started music on hold, class 'default', on Zap/7-1 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '8' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '7' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu' -- Stopped music on hold on Zap/7-1 Oct 15 13:32:58 NOTICE[100377]: chan_mgcp.c:1151 mgcp_fixup: mgcp_fixup(Zap/7-1, Zap/7-1MASQ) -- Swapping 0 for 1 on aaln/[EMAIL PROTECTED] Oct 15 13:32:58 WARNING[14350]: chan_mgcp.c:3033 handle_request: Transfer attempt failed __ Hey There, I had a similar problem running CVS HEAD 02/09/05. An Attended transfer that wasn't completed caused the channel to lockup. with a bunch of: Mar 16 14:12:12 WARNING[8904]: channel.c:523 ast_channel_walk_locked: Avoided deadlock for 'MGCP/aaln/[EMAIL PROTECTED]', 10 retries! messages.. Did you ever find a solution?? Thanks, Brett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk crashes from time to time
(There are other debug modes, but not sure I'd use those to catch a production problem. The one's I know about are primarily intended for development debugging. Other folks might contribute hints here.) This reeks of a deadlock, http://voip-info.org/wiki-Asterisk+deadlock see this HowTo Debug a DeadLock in Asterisk i wrote up eons ago on the wiki http://voip-info.org/wiki-Asterisk+debugging Thanks for your informations, I will try to follow the instructions on debugging asterisk. Since I'm not a programmer, I think I will get some fun with it ;-) Guido ___ 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] Asterisk crashes from time to time
Hello List, we have 3 Asterisk boxes running under Fedora Core 2. Every box hangs/crashes from time to time. These installations are image based, means we made an image from our testserver with an image tool, which is able to manage ext3 partitions and deployed it to different server hardware. These servers run very stable and I could not find any failures in the logs. As these crashes appeared the first time, I thought rebooting these machines by cronjob every night at 04:00 would solve the problems. It seemed to work quite well for a couple of weeks. Today I saw our own asterisk production server crash :-( . These crashes are always the same, asterisk stops responding, the cli does not give any reaction on command input, you have to manually kill -9 all asterisk and moh processes. Asterisk logs are empty. We don't use any isdn/fxs/fxo/e1/t1 cards in these servers. Our connections to PSTN is only made by Patton/Inalp SmartNode Gateways, connected to asterisk via sip protocol. Scince these crashes appear on three servers with different hardware, and the main installation is always the same, I would think there are only two possible sources to find the failure: Operating System Fedora Core 2 Kernel 2.6.8-1.521 Asterisk CVS-HEAD-01/08/05 Has anybody out there similar problems, and if yes, how did he fix them? Is there any working solution, having asterisk control itself perhaps by using a script that drops a test call in /var/spool/asterisk/outgoing and if this call wasn't processed successfull the script stops all running asterisk and moh processes and restarts asterisk? Any help would be appreciated, since I can't get no sleep with these timebombs out there ;-) Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk crashes from time to time
we have 3 Asterisk boxes running under Fedora Core 2. Every box hangs/crashes from time to time. These installations are image based, means we made an image from our testserver with an image tool, which is able to manage ext3 partitions and deployed it to different server hardware. These servers run very stable and I could not find any failures in the logs. As these crashes appeared the first time, I thought rebooting these machines by cronjob every night at 04:00 would solve the problems. It seemed to work quite well for a couple of weeks. Today I saw our own asterisk production server crash :-( . These crashes are always the same, asterisk stops responding, the cli does not give any reaction on command input, you have to manually kill -9 all asterisk and moh processes. Asterisk logs are empty. We don't use any isdn/fxs/fxo/e1/t1 cards in these servers. Our connections to PSTN is only made by Patton/Inalp SmartNode Gateways, connected to asterisk via sip protocol. Scince these crashes appear on three servers with different hardware, and the main installation is always the same, I would think there are only two possible sources to find the failure: Operating System Fedora Core 2 Kernel 2.6.8-1.521 Asterisk CVS-HEAD-01/08/05 Has anybody out there similar problems, and if yes, how did he fix them? Is there any working solution, having asterisk control itself perhaps by using a script that drops a test call in /var/spool/asterisk/outgoing and if this call wasn't processed successfull the script stops all running asterisk and moh processes and restarts asterisk? Far too many variables for anyone to even guess at the root cause. Problem could be related to slight differences in o/s libraries between systems, coding problems within asterisk, etc. There were some issues reported with cvs head in January relative to hangs, etc. Might consider changing /etc/asterisk/logger.conf and add debug to the list. Then after a failure, at least look at /var/log/asterisk/debug messages. For additional info, I'd suggest compiling the code on one of thse machines to see if it complains about missing/inappropriate items. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk crashes from time to time
Rich, Far too many variables for anyone to even guess at the root cause. Problem could be related to slight differences in o/s libraries between systems, coding problems within asterisk, etc. You 're right, it could be every thing There were some issues reported with cvs head in January relative to hangs, etc. Are they reported in the bugtracker, or in the mailing list? Might consider changing /etc/asterisk/logger.conf and add debug to the list. Then after a failure, at least look at /var/log/asterisk/debug messages. Yes, this was the first thing, I did after the crash showed up. I simply forgot to enable it, since this production server ran long time without problems. But now, following murphy's law, the next crash will never happen ;-) For additional info, I'd suggest compiling the code on one of thse machines to see if it complains about missing/inappropriate items. After these machines were setup, we compiled new code on every machine, since we started with an older version of Asterisk in November 2004. The compiling of asterisk did not show me any relevant (?) errors. But I remember there were some statements (Warnings) in the console output of the make process, I didn't understand. Is this output logged in addition to the console in a logfile somewhere? If so, one could examine this output and hopefully get some hints... Thanks for your help Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk crashes from time to time
Inline... Far too many variables for anyone to even guess at the root cause. Problem could be related to slight differences in o/s libraries between systems, coding problems within asterisk, etc. You 're right, it could be every thing There were some issues reported with cvs head in January relative to hangs, etc. Are they reported in the bugtracker, or in the mailing list? Not all for sure. If you watch the -cvs and -user list, you'd see folks with seg faults, etc, and not too long after that you see a change come through -cvs. Sometimes with comments like 'fix silly typo', etc. Given that cvs head is actually development, at any point in time there could easily be various problems (expected). To try to recreate historically whether you caught a cvs head version that had errors is almost impossible. That's why its important to run cvs head in some sort of pre-production test environment before promoting the code into a customer's machine, etc. (That implies beating the hell out of your test environment.) Might consider changing /etc/asterisk/logger.conf and add debug to the list. Then after a failure, at least look at /var/log/asterisk/debug messages. Yes, this was the first thing, I did after the crash showed up. I simply forgot to enable it, since this production server ran long time without problems. But now, following murphy's law, the next crash will never happen ;-) For additional info, I'd suggest compiling the code on one of thse machines to see if it complains about missing/inappropriate items. After these machines were setup, we compiled new code on every machine, since we started with an older version of Asterisk in November 2004. The compiling of asterisk did not show me any relevant (?) errors. But I remember there were some statements (Warnings) in the console output of the make process, I didn't understand. Is this output logged in addition to the console in a logfile somewhere? If so, one could examine this output and hopefully get some hints... The only two (key) log methods that I know of is to run the cli with several -'s, and turn on debugging in the logger.conf file (which may require you to config /etc/syslog.conf to catch them). Then look at /var/log/asterisk/debug after a failure. (There are other debug modes, but not sure I'd use those to catch a production problem. The one's I know about are primarily intended for development debugging. Other folks might contribute hints here.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk crashes from time to time
(There are other debug modes, but not sure I'd use those to catch a production problem. The one's I know about are primarily intended for development debugging. Other folks might contribute hints here.) This reeks of a deadlock, http://voip-info.org/wiki-Asterisk+deadlock see this HowTo Debug a DeadLock in Asterisk i wrote up eons ago on the wiki http://voip-info.org/wiki-Asterisk+debugging ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk crashes my router!?
On Thu, 2004-12-02 at 16:47, David Filion wrote: Hi, Does anybody else have problems like this. I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek Vigour 2600 ADSL router. My * box is configured with a public IP address which is presented on one of the switch ports on the rear of the router. When there is some SIP activity, incoming mainly, towards my * box, the router will lockup after a short period?! Maybe the router can't handle the traffic? If you have a modem before your router, try connecting * right to the modem and using rpppoe. The Vigor 2600 should just do fine, I have 2 Vigor 2600VGi's in use with Asterisk (IAX though) and Handytone's (SIP, not connected to the Asterisk) in use (router PPPoE though, connected to NetSource here in Ireland). Firmware is 2.5.3 (i think Draytek withdrew that again because of GUI problems), but 2.5.2 was fine, too. The router is SIP aware, so actually you shouldn't think much about NAT setup etc, it should work straight away. Have you talked with Draytek about that problem ? Maybe they have heard about it before and have a solution. Besides, is it the UK specific firmware you have loaded ? Slán leat, Martin List-Petersen Dublin, Eire (contact info on -- http://www.marlow.dk/) Yes, try a firmware upgrade. I actually saw a router one time that would lockup if a client behind it ran a trace route ___ 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] Asterisk crashes my router!?
Hi Martin, my router is a vanilla 2600, not the V model, as far as I know it has no special SIP features, other than SIP seeming to crash it when a SIP call is made from the internet to the * box here! :( I mentioned the problem on the draytek forum but I;ve not contacted Draytek themselves per se. One big difference is you are using PPPoE and I'm using PPPoA, unfortunately! I've tried several different firmware, all UK specific, still the same. thanks. Mike On Fri, 03 Dec 2004 23:39:50 +, Martin List-Petersen [EMAIL PROTECTED] wrote: The Vigor 2600 should just do fine, I have 2 Vigor 2600VGi's in use with Asterisk (IAX though) and Handytone's (SIP, not connected to the Asterisk) in use (router PPPoE though, connected to NetSource here in Ireland). Firmware is 2.5.3 (i think Draytek withdrew that again because of GUI problems), but 2.5.2 was fine, too. The router is SIP aware, so actually you shouldn't think much about NAT setup etc, it should work straight away. Have you talked with Draytek about that problem ? Maybe they have heard about it before and have a solution. Besides, is it the UK specific firmware you have loaded ? Slán leat, Martin List-Petersen Dublin, Eire (contact info on -- http://www.marlow.dk/) ___ 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] Asterisk crashes my router!?
On Thu, 2004-12-02 at 16:47, David Filion wrote: Hi, Does anybody else have problems like this. I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek Vigour 2600 ADSL router. My * box is configured with a public IP address which is presented on one of the switch ports on the rear of the router. When there is some SIP activity, incoming mainly, towards my * box, the router will lockup after a short period?! Maybe the router can't handle the traffic? If you have a modem before your router, try connecting * right to the modem and using rpppoe. The Vigor 2600 should just do fine, I have 2 Vigor 2600VGi's in use with Asterisk (IAX though) and Handytone's (SIP, not connected to the Asterisk) in use (router PPPoE though, connected to NetSource here in Ireland). Firmware is 2.5.3 (i think Draytek withdrew that again because of GUI problems), but 2.5.2 was fine, too. The router is SIP aware, so actually you shouldn't think much about NAT setup etc, it should work straight away. Have you talked with Draytek about that problem ? Maybe they have heard about it before and have a solution. Besides, is it the UK specific firmware you have loaded ? Slán leat, Martin List-Petersen Dublin, Eire (contact info on -- http://www.marlow.dk/) ___ 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] Asterisk crashes my router!?
Hi, Does anybody else have problems like this. I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek Vigour 2600 ADSL router. My * box is configured with a public IP address which is presented on one of the switch ports on the rear of the router. When there is some SIP activity, incoming mainly, towards my * box, the router will lockup after a short period?! I've tried upgrading firmware in the router but it still locks up? Thanks Mike ___ 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] Asterisk crashes my router!?
Sounds like it is time for a different router... There are a few routers out there with buggy nat engines... they are fine when you are doing typical nat, but if you are trying to do 1:1 nat... get a good router or make a BSD box to use as a router. I would highly recommend http://m0n0.ch/wall for a great do it yourself router with nice web interface... It does 1:1 nat as well as bridging packet filter... Sean On Thu, 2004-12-02 at 15:54 +, Mike Dent wrote: Hi, Does anybody else have problems like this. I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek Vigour 2600 ADSL router. My * box is configured with a public IP address which is presented on one of the switch ports on the rear of the router. When there is some SIP activity, incoming mainly, towards my * box, the router will lockup after a short period?! I've tried upgrading firmware in the router but it still locks up? Thanks Mike ___ 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 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] Asterisk crashes my router!?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mike Dent Sent: December 2, 2004 10:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk crashes my router!? Hi, Does anybody else have problems like this. I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek Vigour 2600 ADSL router. My * box is configured with a public IP address which is presented on one of the switch ports on the rear of the router. When there is some SIP activity, incoming mainly, towards my * box, the router will lockup after a short period?! I've tried upgrading firmware in the router but it still locks up? Thanks Mike ___ 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 Maybe the router can't handle the traffic? If you have a modem before your router, try connecting * right to the modem and using rpppoe. David ___ 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] Asterisk crashes my router!?
I have exactly the same problem. It occurs with asterisk server based locally or external to the local net. My way around it is to use an external modem and then use a Draytek 2900 for the dual subnet routing. I'm hoping that will solve it as I'm getting dropped calls left right and centre... -- Message: 7 Date: Thu, 2 Dec 2004 11:47:02 -0500 From: David Filion [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk crashes my router!? To: 'Mike Dent' [EMAIL PROTECTED], 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mike Dent Sent: December 2, 2004 10:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk crashes my router!? Hi, Does anybody else have problems like this. I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek Vigour 2600 ADSL router. My * box is configured with a public IP address which is presented on one of the switch ports on the rear of the router. When there is some SIP activity, incoming mainly, towards my * box, the router will lockup after a short period?! I've tried upgrading firmware in the router but it still locks up? Thanks Mike ___ 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 Maybe the router can't handle the traffic? If you have a modem before your router, try connecting * right to the modem and using rpppoe. David ___ 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] Asterisk crashes my router!?
Sean Cook wrote: Sounds like it is time for a different router... There are a few routers out there with buggy nat engines... they are fine when you are doing typical nat, but if you are trying to do 1:1 nat... get a good router or I've been very happy with Netopia 3386-ENTs. make a BSD box to use as a router. I would highly recommend http://m0n0.ch/wall for a great do it yourself router This was so funny that I had to share it: m0n0wall is probably *the first UNIX system that has its boot-time configuration done with PHP*, rather than the usual shell scripts, and that has *the entire system configuration stored in XML format*. There's an excellent reason they're the first: those are both such unbelieveably terrible ideas, especially the PHP init scripts. I would reccomend IPCop, because their designers are a little more grounded. Nick ___ 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] Asterisk crashes my router!?
I've been very happy with Netopia 3386-ENTs. make a BSD box to use as a router. I would highly recommend http://m0n0.ch/wall for a great do it yourself router This was so funny that I had to share it: m0n0wall is probably *the first UNIX system that has its boot-time configuration done with PHP*, rather than the usual shell scripts, and that has *the entire system configuration stored in XML format*. There's an excellent reason they're the first: those are both such unbelieveably terrible ideas, especially the PHP init scripts. Wow, that is one of the most amazing statements I have ever heard... of course you have worked with m0n0wall and have tested it thoroughly to have come to such a conclusion. Of course you have... Exactly what is terrible about php init scripts? Is it that php is a newer language that may not have originally been designed for such a thing? Please educate me... Sean ___ 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] Asterisk crashes my router!?
On 03/12/2004 04:01 Nick Bachmann said the following: There's an excellent reason they're the first: those are both such unbelieveably terrible ideas, especially the PHP init scripts. I would reccomend IPCop, because their designers are a little more would you elaborate why these are terrible ideas ? i'm sure, of course, that you actually used m0n0wall and evaluated it before coming up with that statement. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk crashes my router!?
On Fri, 2004-12-03 at 12:39 +0800, Dinesh Nair wrote: On 03/12/2004 04:01 Nick Bachmann said the following: There's an excellent reason they're the first: those are both such unbelieveably terrible ideas, especially the PHP init scripts. I would reccomend IPCop, because their designers are a little more would you elaborate why these are terrible ideas ? i'm sure, of course, that you actually used m0n0wall and evaluated it before coming up with that statement. [EMAIL PROTECTED]:~$ ls -l /bin/bash -h -rwxr-xr-x 1 root root 652K Nov 11 00:42 /bin/bash [EMAIL PROTECTED]:~$ ldd /bin/bash libncurses.so.5 = /lib/libncurses.so.5 (0x40028000) libdl.so.2 = /lib/libdl.so.2 (0x40067000) libc.so.6 = /lib/libc.so.6 (0x4006a000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x4000) [EMAIL PROTECTED]:~$ ldd /bin/bash|awk '{print $3}'|xargs ls -lHh -rwxr-xr-x 1 root root 88K Oct 13 14:40 /lib/ld-linux.so.2 -rw-r--r-- 1 root root 1.2M Oct 13 14:40 /lib/libc.so.6 -rw-r--r-- 1 root root 9.7K Oct 13 14:40 /lib/libdl.so.2 -rw-r--r-- 1 root root 247K May 27 2004 /lib/libncurses.so.5 Or about a total of 2.2 megs [EMAIL PROTECTED]:~$ ls -l /usr/bin/php4 -h -rwxr-xr-x 1 root root 2.9M Oct 5 03:49 /usr/bin/php4 [EMAIL PROTECTED]:~$ ldd /usr/bin/php4 libcrypt.so.1 = /lib/libcrypt.so.1 (0x40028000) libnsl.so.1 = /lib/libnsl.so.1 (0x40055000) libexpat.so.1 = /usr/lib/libexpat.so.1 (0x4006a000) libedit.so.2 = /usr/lib/libedit.so.2 (0x4008b000) libncurses.so.5 = /lib/libncurses.so.5 (0x400a7000) libpcre.so.3 = /usr/lib/libpcre.so.3 (0x400e6000) libpanel.so.5 = /usr/lib/libpanel.so.5 (0x400f6000) libdb-4.2.so = /usr/lib/libdb-4.2.so (0x400fa000) libbz2.so.1.0 = /usr/lib/libbz2.so.1.0 (0x401d) libz.so.1 = /usr/lib/libz.so.1 (0x401e) libssl.so.0.9.7 = /usr/lib/i686/cmov/libssl.so.0.9.7 (0x401f2000) libresolv.so.2 = /lib/libresolv.so.2 (0x40223000) libm.so.6 = /lib/libm.so.6 (0x40235000) libdl.so.2 = /lib/libdl.so.2 (0x40257000) libc.so.6 = /lib/libc.so.6 (0x4025a000) libcrypto.so.0.9.7 = /usr/lib/i686/cmov/libcrypto.so.0.9.7 (0x4038e000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x4000) [EMAIL PROTECTED]:~$ ldd /usr/bin/php4|awk '{print $3}'|xargs ls -lHh -rwxr-xr-x 1 root root 88K Oct 13 14:40 /lib/ld-linux.so.2 -rw-r--r-- 1 root root 1.2M Oct 13 14:40 /lib/libc.so.6 -rw-r--r-- 1 root root 19K Oct 13 14:40 /lib/libcrypt.so.1 -rw-r--r-- 1 root root 9.7K Oct 13 14:40 /lib/libdl.so.2 -rw-r--r-- 1 root root 132K Oct 13 14:40 /lib/libm.so.6 -rw-r--r-- 1 root root 247K May 27 2004 /lib/libncurses.so.5 -rw-r--r-- 1 root root 72K Oct 13 14:40 /lib/libnsl.so.1 -rw-r--r-- 1 root root 64K Oct 13 14:40 /lib/libresolv.so.2 -rw-r--r-- 1 root root 1006K Nov 14 13:43 /usr/lib/i686/cmov/libcrypto.so.0.9.7 -rw-r--r-- 1 root root 194K Nov 14 13:43 /usr/lib/i686/cmov/libssl.so.0.9.7 -rw-r--r-- 1 root root 61K Nov 24 18:23 /usr/lib/libbz2.so.1.0 -rw-r--r-- 1 root root 857K Aug 21 00:27 /usr/lib/libdb-4.2.so -rw-r--r-- 1 root root 106K Aug 30 17:08 /usr/lib/libedit.so.2 -rw-r--r-- 1 root root 127K Oct 19 19:34 /usr/lib/libexpat.so.1 -rw-r--r-- 1 root root 12K May 27 2004 /usr/lib/libpanel.so.5 -rw-r--r-- 1 root root 63K Mar 12 2004 /usr/lib/libpcre.so.3 -rw-r--r-- 1 root root 66K Oct 30 13:49 /usr/lib/libz.so.1 Or about 7.2 megs. Do you gain enough by using php to explain an extra 5 megs or so over the normal bash. Of course you could go the busybox route and be in at a total of 937k or over 6 megs less executables but a crap load more functionality. So quickly you get the fact that on a minimalistic system such as a firewall, you don't want all those libraries and crap. A true firewall should be so minimal it would easily fit on a floppy image and be read only so as not to be very exploitable. And for a non technical argument, the use of php for the init scripts smacks of someone who knew php and thought they would reinvent the wheel(firewall) with the only technology they knew how to use. If true, I would worry about security. -- 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
Re: [Asterisk-Users] Asterisk crashes my router!?
ahh, the famed steve critchfield has honoured me. On 03/12/2004 13:21 Steven Critchfield said the following: [EMAIL PROTECTED]:~$ ldd /bin/bash libncurses.so.5 = /lib/libncurses.so.5 (0x40028000) libdl.so.2 = /lib/libdl.so.2 (0x40067000) libc.so.6 = /lib/libc.so.6 (0x4006a000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x4000) [EMAIL PROTECTED]:~$ ldd /bin/bash|awk '{print $3}'|xargs ls -lHh -rwxr-xr-x 1 root root 88K Oct 13 14:40 /lib/ld-linux.so.2 -rw-r--r-- 1 root root 1.2M Oct 13 14:40 /lib/libc.so.6 -rw-r--r-- 1 root root 9.7K Oct 13 14:40 /lib/libdl.so.2 -rw-r--r-- 1 root root 247K May 27 2004 /lib/libncurses.so.5 Or about a total of 2.2 megs interesting, because on m0n0wall, #ls -al /usr/local/bin/php -r-xr-xr-x 1 root wheel 1060380 Jul 18 15:57 php #ldd /usr/local/bin/php php: libcrypt.so.2 = /usr/lib/libcrypt.so.2 (0x2815d000) libm.so.2 = /usr/lib/libm.so.2 (0x28176000) libc.so.4 = /usr/lib/libc.so.4 (0x28191000) # ls -al /usr/lib/libcrypt.so.2 /usr/lib/libm.so.2 /usr/lib/libc.so.4 -r--r--r-- 1 root wheel 580572 Jun 24 04:03 /usr/lib/libc.so.4 -r--r--r-- 1 root wheel 28432 Jun 24 04:03 /usr/lib/libcrypt.so.2 -r--r--r-- 1 root wheel 117024 Jun 24 04:03 /usr/lib/libm.so.2 which makes it 1.7MB by my calculations (1024*1024 bytes per MB). Or about 7.2 megs. Do you gain enough by using php to explain an extra 5 megs or so over the normal bash. Of course you could go the busybox so i guess, the question is, do you gain enough by using bash to warrant the extra 30% or so ? So quickly you get the fact that on a minimalistic system such as a firewall, you don't want all those libraries and crap. A true firewall perhaps for a CLI based system. take a look at picobsd for something which fits off a single 1.44MB floppy. however, m0n0wall also provides VPN, captive portal, traffic shaper, DHCP and NAT functionality /with/ a web-based GUI for people who're /unfamiliar/ with the CLI. wheel(firewall) with the only technology they knew how to use. If true, I would worry about security. once again, in what manner ? remember, we're discussing the init scripts. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ 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] Asterisk crashes with Unicall
Hi, For the last 40 days i've been using Unicall on an Asterisk connected to an Ericsson MD-110 PBX. It was working fine for two weeks when there were just some random calls but for the last two weeks when the load increased to between 5 and 10 simultaneous calls the system became unreliable with 2 main problems: 1- Some dropped calls when the call comes from Unicall: Unicall - IAX/SIP. When it comes from Zap (E1 PRI) there is no problem: Zap - IAX/SIP. 2- Asterisk crashes 2 or 3 times a day. Always when there is some Unicall channel active. To be sure that the crashes are Unicall related I created an test enviroment: 2 servers with the same configuration: - P4 2.8Ghz - 512MB - 1 Digium E100P (connected with each other using a E1 cross cable) The test was: using an .call file to start a call from 1 server to the other on an extension that dial to the first server, that dial to the other and so on... until there is no more channels available. The result: the calls start ringing in both servers until there is no more channels free, then they start to timeout and hangup. Until here there is no problem, but then suddenly one of the Asterisk servers crashes. Sometimes the server that initiated the calls, sometimes the other, there is no pattern. (I repeated the test several times and one time both Asterisk crashed). If I change the signalling to E1 PRI and make the same test there is no problem (calls ring until no more channels are available and timeout after some time). Some messages from the Asterisk that crashed follows below (Got only the last 200 lines, the complete log is 1800 lines / 192 Kb, too big for posting here). Is there some debugging info i can extract from this test and post here to help ? Thanks, Leonardo zaptel.conf: span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 unicall.conf: [channels] language=br context=principal_in rxwink=300 usecallerid=yes hidecallerid=no usecallingpres=yes callprogress=no restrictcid=no immediate=no callwaitingcallerid=no threewaycalling=no transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 protocolclass=mfcr2 protocolvariant=br,4,4 protocolend=cpe #co on the other server group=1 callerid=asreceived context=principal_in channel=1-15 channel=17-31 extensions.conf: [principal_in] exten = _.,1,SetCallerId() exten = _.,2,Dial(UniCall/g1/${EXTEN},600) Call file: Channel: UniCall/g1/ Callerid: MaxRetries: 0 RetryTime: 600 WaitTime: 600 Context: principal_in Extension: 777 Priority: 1 Core file: Core was generated by `/usr/sbin/asterisk -fg'. Program terminated with signal 11, Segmentation fault. #0 0x407fa684 in ?? () No debugging symbols on asterisk binary cause it was installed from the RPM and the building process strips symbols. I can install other binary with debugging if it helps... Asterisk messages: Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Detected Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Making a new call with CRN 32769 Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx bits 0xD [2/ 2/101/ 0] Nov 19 09:41:25 WARNING[7175]: UC event Detected Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 on [2/ 2/101/ 0] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 on [2/ 2/101/ 0] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 off [2/ 2/102/101] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 off [2/ 2/102/101] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 on [2/ 2/102/101] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 on [2/ 2/102/101] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 off [2/ 2/102/101] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 off [2/ 2/102/101] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 on [2/ 2/102/101] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 on [2/ 2/102/101] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 off [2/ 2/102/101] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 1 off [2/ 2/102/101] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 on [2/ 2/102/101] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 5 on [2/ 2/102/101] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 0 off [2/ 2/102/105] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 5 off [2/ 2/102/105] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 6 on [2/ 2/102/105] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 5 on [2/ 2/102/105] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Rx tone 6 off [2/ 2/102/103] Nov 19 09:41:25 WARNING[7175]: UniCall: mfcr2 Tx tone 5 off [2/ 2/102/103] Nov 19 09:41:26 WARNING[7175]: UniCall: mfcr2 Rx tone 6 on [2/ 2/102/103] Nov 19 09:41:26 WARNING[7175]: UniCall: mfcr2 Tx tone 5 on [2/ 2/102/103] Nov 19 09:41:26 WARNING[7175]: UniCall: mfcr2 Rx tone 6 off [2/ 2/102/103] Nov 19 09:41:26 WARNING[7175]: UniCall: mfcr2 Tx tone 5 off [2/ 2/102/103] Nov 19
[Asterisk-Users] Asterisk crashes after call when running as non-root, bug???
Hi all, I am using Debian Sarge (2.6) with ISDN4Linux. If I run asterisk as root everyting is OK. If I run asterisk as the user asterisk, the programm crashes after answering a call. No message in the asterisk logs but the /var/log/messages says: Nov 12 20:42:01 localhost kernel: isdn: HiSax1,ch0 cause: E0010 Is this a bug or is this a known feature that can be solved by properly configuring asterisk? TIA Joost ___ 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] Asterisk crashes on special Transfer with MGCP/ATA 186
Hi all, i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco ATA-186 3.1.1 atamgcp We are used to make an special ;) blind transfer like (Flash)Number(Hangup before anyone answers or ring). Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp If one waits until the last one rings, then hangup, everything is fine. If one waits until the last one answers, then hangup, everything is fine, too. Any hints? mgcp debug on: -- Executing AGI(Zap/7-1, nuller.agi) in new stack -- Launched AGI Script /home/kpj/pbx/var/lib/asterisk/agi-bin/nuller.agi -- Accepting call from '01635571857' to '8551' on channel 0/1, span 3 -- AGI Script nuller.agi completed, returning 0 -- Executing Dial(Zap/7-1, MGCP/aaln/[EMAIL PROTECTED]||) in new stack -- MGCP mgcp_request(aaln/[EMAIL PROTECTED]) -- MGCP cw: 0, dnd: 0, so: 0, sno: 0 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Called aaln/[EMAIL PROTECTED] -- MGCP/aaln/[EMAIL PROTECTED] is ringing -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP/aaln/[EMAIL PROTECTED] answered Zap/7-1 gw-bzo*CLI mgcp debug on Usage: mgcp debug Enables dumping of MGCP packets for debugging purposes -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf' -- Swapping 1 for 0 on aaln/[EMAIL PROTECTED] -- MGCP Muting 1 on aaln/[EMAIL PROTECTED] -- Started music on hold, class 'default', on Zap/7-1 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '8' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '7' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu' -- Stopped music on hold on Zap/7-1 Oct 15 13:32:58 NOTICE[100377]: chan_mgcp.c:1151 mgcp_fixup: mgcp_fixup(Zap/7-1, Zap/7-1MASQ) -- Swapping 0 for 1 on aaln/[EMAIL PROTECTED] Oct 15 13:32:58 WARNING[14350]: chan_mgcp.c:3033 handle_request: Transfer attempt failed ___ 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] Asterisk crashes when trying to load G.729module.
Try to install the new codec code that is available in ftp://ftp.digium.com/pub/telephony/asterisk/g729/new_codec_binary/codec_g729b.so place it in /usr/lib/asterisk/modules and restart asterisk (or try to start it). There is also a new command available g.729 show license usage and a few fixes to the code. Write back about the results. regards Martin On Sun, 20 Jul 2003, Anton Tinchev wrote: Before few days i bought few g.729 licenses. When i try to load the codec, asterisk crahses. I tried with and without oh323 module, same result: -- Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 -- Here the ldd result: -- [EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so libc.so.6 = /lib/libc.so.6 (0x40039000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000) Version information: /usr/lib/asterisk/modules/codec_g729b.so: libc.so.6 (GLIBC_2.1.3) = /lib/libc.so.6 libc.so.6 (GLIBC_2.2) = /lib/libc.so.6 libc.so.6 (GLIBC_2.1) = /lib/libc.so.6 libc.so.6 (GLIBC_2.0) = /lib/libc.so.6 /lib/libc.so.6: ld-linux.so.2 (GLIBC_2.1) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_2.0) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_PRIVATE) = /lib/ld-linux.so.2 --- ___ 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] Asterisk crashes when trying to load G.729 module.
You have to run a console with the G.729 due to the voice age library lameness. We run safe_asterisk with a TTY and it seems to be fine. Jeremy McNamara [EMAIL PROTECTED] wrote: Try launching asterisk like this: screen -d -m asterisk -vvvcn Aparently there is some bug in the codec. - Justin On Sun, 20 Jul 2003, Anton Tinchev wrote: Before few days i bought few g.729 licenses. When i try to load the codec, asterisk crahses. I tried with and without oh323 module, same result: -- Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 -- Here the ldd result: -- [EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so libc.so.6 = /lib/libc.so.6 (0x40039000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000) Version information: /usr/lib/asterisk/modules/codec_g729b.so: libc.so.6 (GLIBC_2.1.3) = /lib/libc.so.6 libc.so.6 (GLIBC_2.2) = /lib/libc.so.6 libc.so.6 (GLIBC_2.1) = /lib/libc.so.6 libc.so.6 (GLIBC_2.0) = /lib/libc.so.6 /lib/libc.so.6: ld-linux.so.2 (GLIBC_2.1) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_2.0) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_PRIVATE) = /lib/ld-linux.so.2 --- ___ 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] Asterisk crashes when trying to load G.729 module.
Is it stable enought? I mean around 30-40 Incoming SIP connections. Or i must trash the cisco and put Asteriisk/Digium/speex box? Jeremy McNamara wrote: You have to run a console with the G.729 due to the voice age library lameness. We run safe_asterisk with a TTY and it seems to be fine. Jeremy McNamara [EMAIL PROTECTED] wrote: Try launching asterisk like this: screen -d -m asterisk -vvvcn Aparently there is some bug in the codec. - Justin On Sun, 20 Jul 2003, Anton Tinchev wrote: Before few days i bought few g.729 licenses. When i try to load the codec, asterisk crahses. I tried with and without oh323 module, same result: -- Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 -- Here the ldd result: -- [EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so libc.so.6 = /lib/libc.so.6 (0x40039000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000) Version information: /usr/lib/asterisk/modules/codec_g729b.so: libc.so.6 (GLIBC_2.1.3) = /lib/libc.so.6 libc.so.6 (GLIBC_2.2) = /lib/libc.so.6 libc.so.6 (GLIBC_2.1) = /lib/libc.so.6 libc.so.6 (GLIBC_2.0) = /lib/libc.so.6 /lib/libc.so.6: ld-linux.so.2 (GLIBC_2.1) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_2.0) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_PRIVATE) = /lib/ld-linux.so.2 --- ___ 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] Asterisk crashes when trying to load G.729 module.
This is generally indicates a problem with the licensing process (which is severely flawed and full of bugs) on your server... Did you make it through the registration process OK? Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Tinchev Sent: Sunday, July 20, 2003 12:18 AM To: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk crashes when trying to load G.729 module. Before few days i bought few g.729 licenses. When i try to load the codec, asterisk crahses. I tried with and without oh323 module, same result: -- Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 -- Here the ldd result: -- [EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so libc.so.6 = /lib/libc.so.6 (0x40039000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000) Version information: /usr/lib/asterisk/modules/codec_g729b.so: libc.so.6 (GLIBC_2.1.3) = /lib/libc.so.6 libc.so.6 (GLIBC_2.2) = /lib/libc.so.6 libc.so.6 (GLIBC_2.1) = /lib/libc.so.6 libc.so.6 (GLIBC_2.0) = /lib/libc.so.6 /lib/libc.so.6: ld-linux.so.2 (GLIBC_2.1) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_2.0) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_PRIVATE) = /lib/ld-linux.so.2 --- ___ 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] Asterisk crashes when trying to load G.729 module.
Before few days i bought few g.729 licenses. When i try to load the codec, asterisk crahses. I tried with and without oh323 module, same result: -- Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 -- Here the ldd result: -- [EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so libc.so.6 = /lib/libc.so.6 (0x40039000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000) Version information: /usr/lib/asterisk/modules/codec_g729b.so: libc.so.6 (GLIBC_2.1.3) = /lib/libc.so.6 libc.so.6 (GLIBC_2.2) = /lib/libc.so.6 libc.so.6 (GLIBC_2.1) = /lib/libc.so.6 libc.so.6 (GLIBC_2.0) = /lib/libc.so.6 /lib/libc.so.6: ld-linux.so.2 (GLIBC_2.1) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_2.0) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_PRIVATE) = /lib/ld-linux.so.2 --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users