Re: [asterisk-users] Realtime problem host='dynamic' in 1.2.26.1
Developers and maintainers, any information? // T Torbjörn Abrahamsson wrote: Hello! We are using the 1.2 branch, and upgraded to 1.2.26.1. We ran into some problems when using realtime for peers. We connect the PBX to a sip peer at an ITSP, and when we try to dial the peer we get: Jan 23 09:02:07 VERBOSE[2236] logger.c: -- Executing Dial(SIP/dev02-08c36f28, SIP/[EMAIL PROTECTED]||W) in new stack Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Everything is fine. Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sippeers WHERE name = '989800-out' AND host = 'dynamic' Jan 23 09:02:07 WARNING[2236] chan_sip.c: No such host: 989800-out Jan 23 09:02:07 NOTICE[2236] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Jan 23 09:02:07 VERBOSE[2236] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Jan 23 09:02:07 DEBUG[2236] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. I looked in the archives and found this thread: http://lists.digium.com/pipermail/asterisk-users/2007-December/202616.html Here the same problem is discussed for the 1.4 branch, and the result is that the problem should be fixed. But this is still a problem in 1.2 branch. Will this be corrected in a new release, or is this not considered a security fix and hence ignored? Actually isn't this a fix for a security fix... BR, Torbjörn Abrahamsson ___ -- 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] Realtime problem host='dynamic' in 1.2.26.1
Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sippeers WHERE name = '989800-out' AND host = 'dynamic' Quite obvious .. doest sippeers have that row ? On Jan 24, 2008 6:04 PM, Torbjörn Abrahamsson [EMAIL PROTECTED] wrote: Developers and maintainers, any information? // T Torbjörn Abrahamsson wrote: Hello! We are using the 1.2 branch, and upgraded to 1.2.26.1. We ran into some problems when using realtime for peers. We connect the PBX to a sip peer at an ITSP, and when we try to dial the peer we get: Jan 23 09:02:07 VERBOSE[2236] logger.c: -- Executing Dial(SIP/dev02-08c36f28, SIP/[EMAIL PROTECTED]||W) in new stack Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Everything is fine. Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sippeers WHERE name = '989800-out' AND host = 'dynamic' Jan 23 09:02:07 WARNING[2236] chan_sip.c: No such host: 989800-out Jan 23 09:02:07 NOTICE[2236] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Jan 23 09:02:07 VERBOSE[2236] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Jan 23 09:02:07 DEBUG[2236] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. I looked in the archives and found this thread: http://lists.digium.com/pipermail/asterisk-users/2007-December/202616.html Here the same problem is discussed for the 1.4 branch, and the result is that the problem should be fixed. But this is still a problem in 1.2branch. Will this be corrected in a new release, or is this not considered a security fix and hence ignored? Actually isn't this a fix for a security fix... BR, Torbjörn Abrahamsson ___ -- 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
Re: [asterisk-users] Realtime problem host='dynamic' in 1.2.26.1
Yes, it does... This is not a problem of usage. This mail should probably have been sent to -dev instead, as it clearly is a bug. If you take a look at the link I provided you will see that this is indeed a bug, and it has been fixed in 1.4 branch in 1.4.17. The problem is that 1.2 is in security fixes only-mode, and thereby this should not be fixed. My question is, that as this is a bug that occurs as a result of a broken security fix, will a fix for this be released anyway? I have backported the changes to chan_sip.c 1.4-fix for this to 1.2.26.1, so I do have a working solution but I hope to be able to use an official release. // T Jaswinder Singh wrote: Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sippeers WHERE name = '989800-out' AND host = 'dynamic' Quite obvious .. doest sippeers have that row ? On Jan 24, 2008 6:04 PM, Torbjörn Abrahamsson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Developers and maintainers, any information? // T Torbjörn Abrahamsson wrote: Hello! We are using the 1.2 branch, and upgraded to 1.2.26.1 http://1.2.26.1. We ran into some problems when using realtime for peers. We connect the PBX to a sip peer at an ITSP, and when we try to dial the peer we get: Jan 23 09:02:07 VERBOSE[2236] logger.c: -- Executing Dial(SIP/dev02-08c36f28, SIP/[EMAIL PROTECTED]||W) in new stack Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Everything is fine. Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sippeers WHERE name = '989800-out' AND host = 'dynamic' Jan 23 09:02:07 WARNING[2236] chan_sip.c: No such host: 989800-out Jan 23 09:02:07 NOTICE[2236] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Jan 23 09:02:07 VERBOSE[2236] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Jan 23 09:02:07 DEBUG[2236] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. I looked in the archives and found this thread: http://lists.digium.com/pipermail/asterisk-users/2007-December/202616.html Here the same problem is discussed for the 1.4 branch, and the result is that the problem should be fixed. But this is still a problem in 1.2 branch. Will this be corrected in a new release, or is this not considered a security fix and hence ignored? Actually isn't this a fix for a security fix... BR, Torbjörn Abrahamsson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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 ___ -- 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] Realtime problem host='dynamic' in 1.2.26.1
On Thursday 24 January 2008 08:54:03 Jaswinder Singh wrote: Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sippeers WHERE name = '989800-out' AND host = 'dynamic' Quite obvious .. doest sippeers have that row ? Or download 1.2.26.2. -- Tilghman ___ -- 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] Realtime problem host='dynamic' in 1.2.26.1
Huh? As far as I can see, the latest 1.2-release is 1.2.26.1, at least in tar-balls... Hmm.. OK, looking in svn I now see the 1.2.26.2 release... Shouldn't it be in tar-balls as well? Another hmmm Interesting... After looking at the download page again now, about 5 minutes after my lastest try, the file is there... Very interesting... Ah well... All is good then! // T -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: den 24 januari 2008 16:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime problem host='dynamic' in 1.2.26.1 On Thursday 24 January 2008 08:54:03 Jaswinder Singh wrote: Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sippeers WHERE name = '989800-out' AND host = 'dynamic' Quite obvious .. doest sippeers have that row ? Or download 1.2.26.2. -- Tilghman ___ -- 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] realtime problem
Try looking at this link: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf +sorting Bobby ___ --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] Realtime problem
On 16:57, Thu 22 Jun 06, Benjamin Stocker wrote: Hi This works fine in extensions.conf: exten = _0X./100,1,Dial(SIP/[EMAIL PROTECTED]) exten = _0X./200,1,Dial(SIP/[EMAIL PROTECTED]) This will just use different SIP channels for different Caller ID's. If I write the same to a realtime table, Asterisk always uses sipout-a, no matter what Caller ID is used. That will be the case with static configs too, because the argument to Dial is the same in both cases -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --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] Realtime problem
2006/6/22, Michiel van Baak [EMAIL PROTECTED]: On 16:57, Thu 22 Jun 06, Benjamin Stocker wrote: Hi This works fine in extensions.conf: exten = _0X./100,1,Dial(SIP/[EMAIL PROTECTED]) exten = _0X./200,1,Dial(SIP/[EMAIL PROTECTED] ) This will just use different SIP channels for different Caller ID's. If I write the same to a realtime table, Asterisk always uses sipout-a, no matter what Caller ID is used.That will be the case with static configs too, because the argument to Dial is the same in both casesThat was a typo. Sorry, the second line reads:exten = _0X./200,1,Dial(SIP/[EMAIL PROTECTED]) ___ --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] RealTime problem with sipusers accounts
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I need to add and remove Sip accounts in realtime. What's the best way at the moment to do that? * Add/remove the user into the sip.conf and execute asterisk -x 'sip reload' ? Thanks for help Marco Kevin P. Fleming schrieb: Marco Balmer wrote: Server1 acts as a SIP Client only. Server2 should act as a SIP-Server with the sip_buddies table on the MySQL-Server. But this is not currently implemented. There is a patch in the bug tracker that will help move in this direction, but it's only a start, there are many more issues that need to be resolved for this to work properly. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDUpHq8JLvhlgYtaoRAqOEAKCXsI3TLL23DDpzzMZi3cno4xqOTQCfUzX2 GCaR660+WeEHV/HayHwm4qY= =Sm3A -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] RealTime problem with sipusers accounts
Marco Balmer wrote: Any ideas or hints? Yes. Whatever documentation told you that you could share a Realtime SIP peer database between two Asterisk servers was in error (or at least very incomplete). There are ways to do it right now, but it's not trivial and does not provide all the functionality that someone would want from such an arrangement. ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] RealTime problem with sipusers accounts
Hello On Fri, 14 Oct 2005 01:25:20 -0500, Kevin P. Fleming wrote Marco Balmer wrote: Any ideas or hints? Yes. Whatever documentation told you that you could share a Realtime SIP peer database between two Asterisk servers was in error (or at least very incomplete). Server1 acts as a SIP Client only. Server2 should act as a SIP-Server with the sip_buddies table on the MySQL-Server. Thanks Marco ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] RealTime problem with sipusers accounts
Marco Balmer wrote: Server1 acts as a SIP Client only. Server2 should act as a SIP-Server with the sip_buddies table on the MySQL-Server. But this is not currently implemented. There is a patch in the bug tracker that will help move in this direction, but it's only a start, there are many more issues that need to be resolved for this to work properly. ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] Realtime Problem = Segmentation faults Seems to be asterisk
Jose R. Ortiz wrote: Greg Boehnlein wrote: On Fri, 18 Mar 2005, Jose R. Ortiz Ubarri wrote: Jose R. Ortiz Ubarri wrote: Hi: I had asterisk with RealTime database working perfectly in a RH 9.0 machine. I used the sip cache so I even had MWI working. The problem is that I decided to move to Fedora Core 3. I installed the lastets cvs version of asterisk and the RealTime addon from asterisk-addons. I at first had the problems with the kernel and the zaptel driver but all that was solved with the configuration from the Asterisk Wiki. Then when I moved my configuration to the new asterisk server and configured the RealTime addon it falls in a Segmentation fault. If I do not load the res_config_mysql.so (edited at modules.conf) then asterisks runs without any problem. But if I load the module from boot or from the asterisk command load res_config_mysql.so then I get the Segmentation fault again. I'm not sure what the problem is. Is it a Fedora Core 3 problem, or an Asterisk latest version problem? I don't think it is a configuration problem because I just used the same configuration I had before. The only diferences may be the OS and probably the asterisk version that is only one week newer than the one I was running in the old asterisk server, so I'm probably even running the same version of asterisk in both machines. Any advise? Someone else have a similar configuration working with Fedora Core 3? Thanks in advance, Debugging the code and as you can see in the backtrace the problem is that it is receiving a Null variable (name) and then making the comparison. Is it an asterisk bug? What asterisk should do if the variable name received is NULL? Has this been entered into the Bug Tracker? If so, what bug number was it assigned? The bug id is 0003814. I jumped the category drop down and it says to be a Codec Handling Problem. I couldn't edit it... -- JO ___ 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 This bug is fixed in the CVS Head -- Jose R. Ortiz Ubarri (CHEO), CS System Administrator / Programmer High Performance Computing facility - UPR Email: [EMAIL PROTECTED]|[EMAIL PROTECTED] Phone: 787-758-3054 Fax: 787-758-3058 ___ 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] Realtime Problem = Segmentation faults Seems to be asterisk
Greg Boehnlein wrote: On Fri, 18 Mar 2005, Jose R. Ortiz Ubarri wrote: Jose R. Ortiz Ubarri wrote: Hi: I had asterisk with RealTime database working perfectly in a RH 9.0 machine. I used the sip cache so I even had MWI working. The problem is that I decided to move to Fedora Core 3. I installed the lastets cvs version of asterisk and the RealTime addon from asterisk-addons. I at first had the problems with the kernel and the zaptel driver but all that was solved with the configuration from the Asterisk Wiki. Then when I moved my configuration to the new asterisk server and configured the RealTime addon it falls in a Segmentation fault. If I do not load the res_config_mysql.so (edited at modules.conf) then asterisks runs without any problem. But if I load the module from boot or from the asterisk command load res_config_mysql.so then I get the Segmentation fault again. I'm not sure what the problem is. Is it a Fedora Core 3 problem, or an Asterisk latest version problem? I don't think it is a configuration problem because I just used the same configuration I had before. The only diferences may be the OS and probably the asterisk version that is only one week newer than the one I was running in the old asterisk server, so I'm probably even running the same version of asterisk in both machines. Any advise? Someone else have a similar configuration working with Fedora Core 3? Thanks in advance, Debugging the code and as you can see in the backtrace the problem is that it is receiving a Null variable (name) and then making the comparison. Is it an asterisk bug? What asterisk should do if the variable name received is NULL? Has this been entered into the Bug Tracker? If so, what bug number was it assigned? The bug id is 0003814. I jumped the category drop down and it says to be a Codec Handling Problem. I couldn't edit it... -- JO ___ 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] Realtime Problem = Segmentation faults Seems to be asterisk
On Fri, 18 Mar 2005, Jose R. Ortiz Ubarri wrote: Jose R. Ortiz Ubarri wrote: Hi: I had asterisk with RealTime database working perfectly in a RH 9.0 machine. I used the sip cache so I even had MWI working. The problem is that I decided to move to Fedora Core 3. I installed the lastets cvs version of asterisk and the RealTime addon from asterisk-addons. I at first had the problems with the kernel and the zaptel driver but all that was solved with the configuration from the Asterisk Wiki. Then when I moved my configuration to the new asterisk server and configured the RealTime addon it falls in a Segmentation fault. If I do not load the res_config_mysql.so (edited at modules.conf) then asterisks runs without any problem. But if I load the module from boot or from the asterisk command load res_config_mysql.so then I get the Segmentation fault again. I'm not sure what the problem is. Is it a Fedora Core 3 problem, or an Asterisk latest version problem? I don't think it is a configuration problem because I just used the same configuration I had before. The only diferences may be the OS and probably the asterisk version that is only one week newer than the one I was running in the old asterisk server, so I'm probably even running the same version of asterisk in both machines. Any advise? Someone else have a similar configuration working with Fedora Core 3? Thanks in advance, Debugging the code and as you can see in the backtrace the problem is that it is receiving a Null variable (name) and then making the comparison. Is it an asterisk bug? What asterisk should do if the variable name received is NULL? Has this been entered into the Bug Tracker? If so, what bug number was it assigned? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Realtime Problem = Segmentation faults Seems to be asterisk
Jose R. Ortiz Ubarri wrote: Hi: I had asterisk with RealTime database working perfectly in a RH 9.0 machine. I used the sip cache so I even had MWI working. The problem is that I decided to move to Fedora Core 3. I installed the lastets cvs version of asterisk and the RealTime addon from asterisk-addons. I at first had the problems with the kernel and the zaptel driver but all that was solved with the configuration from the Asterisk Wiki. Then when I moved my configuration to the new asterisk server and configured the RealTime addon it falls in a Segmentation fault. If I do not load the res_config_mysql.so (edited at modules.conf) then asterisks runs without any problem. But if I load the module from boot or from the asterisk command load res_config_mysql.so then I get the Segmentation fault again. I'm not sure what the problem is. Is it a Fedora Core 3 problem, or an Asterisk latest version problem? I don't think it is a configuration problem because I just used the same configuration I had before. The only diferences may be the OS and probably the asterisk version that is only one week newer than the one I was running in the old asterisk server, so I'm probably even running the same version of asterisk in both machines. Any advise? Someone else have a similar configuration working with Fedora Core 3? Thanks in advance, Debugging the code and as you can see in the backtrace the problem is that it is receiving a Null variable (name) and then making the comparison. Is it an asterisk bug? What asterisk should do if the variable name received is NULL? -- Jose R. Ortiz Ubarri (CHEO), CS System Administrator / Programmer High Performance Computing facility - UPR Email: [EMAIL PROTECTED]|[EMAIL PROTECTED] Phone: 787-758-3054 Fax: 787-758-3058 ___ 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] Realtime Problem = Segmentation faults Seemsto be asterisk
Jose R. Ortiz Ubarri wrote: Debugging the code and as you can see in the backtrace the problem is that it is receiving a Null variable (name) and then making the comparison. Is it an asterisk bug? What asterisk should do if the variable name received is NULL? Our chan_sip.c are still not synch'd. I can't help if I don't have the right line numbers. What version of chan_sip are you using? Check inside CVS/Entries and I'll make sure I have the same ver. Then do a make clean; make and reinstall and produce the crash and send the backtrace again. If you can also send the relevant entries in your database that relate to this SIP user. -Matthew ___ 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] Realtime Problem = Segmentation faults Seemsto be asterisk
Matthew: I sent more information and more opinions to the Dev and user list. My chan_sip.c file is from /chan_sip.c/1.674/Thu Mar 17 16:11:19 2005// Thanks Matthew for your help! -- JO Matthew Boehm wrote: Jose R. Ortiz Ubarri wrote: Debugging the code and as you can see in the backtrace the problem is that it is receiving a Null variable (name) and then making the comparison. Is it an asterisk bug? What asterisk should do if the variable name received is NULL? Our chan_sip.c are still not synch'd. I can't help if I don't have the right line numbers. What version of chan_sip are you using? Check inside CVS/Entries and I'll make sure I have the same ver. Then do a make clean; make and reinstall and produce the crash and send the backtrace again. If you can also send the relevant entries in your database that relate to this SIP user. -Matthew ___ 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 -- Jose R. Ortiz Ubarri (CHEO), CS System Administrator / Programmer High Performance Computing facility - UPR Email: [EMAIL PROTECTED]|[EMAIL PROTECTED] Phone: 787-758-3054 Fax: 787-758-3058 ___ 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] Realtime Problem = Segmentation faults Seemstobe asterisk
Jose R. Ortiz Ubarri wrote: Matthew: I sent more information and more opinions to the Dev and user list. My chan_sip.c file is from /chan_sip.c/1.674/Thu Mar 17 16:11:19 2005// Oh poo. I forgot that I had patched my chan_sip with Kevin's RPID patch to test it for him. It seems that you found the path that the logic is going. I will read it for more detail. -Matthew ___ 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] Realtime Problem = Segmentation faults
Jose R. Ortiz Ubarri wrote: But if I load the module from boot or from the asterisk command load res_config_mysql.so then I get the Segmentation fault again. Where is your backtrace? I don't see a backtrace anywhere. Hi. My phone isn't working but I'm not going to let you see what I did to cause it to stop working. Send backtrace from the core file when it crashes. -Matthew ___ 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] Realtime Problem = Segmentation faults
Hi: A very nice guy asked me for a trace: (I hope this is what I was asked for) from /var/log/messages: Mar 17 14:04:23 NOTICE[24649]: Registered Config Engine mysql Mar 17 14:04:23 WARNING[24649]: Unable to get our IP address, Skinny disabled Mar 17 14:04:57 NOTICE[24666]: Registered Config Engine mysql Mar 17 14:04:58 WARNING[24666]: Unable to get our IP address, Skinny disabled Mar 17 14:07:48 NOTICE[24696]: Registered Config Engine mysql Mar 17 14:07:48 WARNING[24696]: Unable to get our IP address, Skinny disabled from the output of asterisk -c: Asterisk Ready CLI Segmentation Fault From gdb:Starting program: /usr/sbin/asterisk -f [Thread debugging using libthread_db enabled] [New Thread -151050560 (LWP 24772)] [New Thread -151053392 (LWP 24775)] [Thread -151053392 (LWP 24775) exited] [New Thread -151053392 (LWP 24776)] [New Thread -151376976 (LWP 24777)] [New Thread -151782480 (LWP 24778)] Mar 17 14:13:20 NOTICE[24772]: config.c:847 ast_config_engine_register: Registered Config Engine mysql [New Thread -152441936 (LWP 24780)] [New Thread -154137680 (LWP 24781)] [New Thread -154567760 (LWP 24782)] [New Thread -154862672 (LWP 24783)] Mar 17 14:13:21 WARNING[24772]: chan_skinny.c:2904 reload_config: Unable to get our IP address, Skinny disabled [New Thread -155186256 (LWP 24784)] [New Thread -156058704 (LWP 24785)] [New Thread -156439632 (LWP 24786)] [New Thread -156812368 (LWP 24787)] [New Thread -157078608 (LWP 24788)] [New Thread -159388752 (LWP 24789)] [Thread -159388752 (LWP 24789) exited] [New Thread -159388752 (LWP 24790)] [Thread -159388752 (LWP 24790) exited] [New Thread -159654992 (LWP 24791)] [Thread -159654992 (LWP 24791) exited] [New Thread -159654992 (LWP 24792)] [Thread -159654992 (LWP 24792) exited] Program received signal SIGSEGV, Segmentation fault. [Switching to Thread -152441936 (LWP 24780)] 0x007642b8 in strcasecmp () from /lib/tls/libc.so.6 (gdb) backtrace #0 0x007642b8 in strcasecmp () from /lib/tls/libc.so.6 #1 0xf6eb58c0 in build_peer (name=0x0, v=0x95fa370, realtime=0) at chan_sip.c:9255 #2 0xf6eb67b0 in find_peer (peer=0x0, sin=0x9642fd4, realtime=1) at chan_sip.c:1222 #3 0xf6ebea77 in check_user_full (p=0x9642e78, req=0xf6e9bb50, cmd=0xf6e9bd64 SUBSCRIBE, uri=0xf6e9bd6e sip:[EMAIL PROTECTED]:5060, reliable=0, sin=0xf6e9bb40, ignore=0, mailbox=0xf6e920a0 , mailboxlen=106) at chan_sip.c:5844 #4 0xf6ec3129 in handle_request (p=0x9642e78, req=0xf6e9bb50, sin=0xf6e9bb40, recount=0x6a, nounlock=0xf6e9b9c8) at chan_sip.c:8384 #5 0xf6ec5281 in sipsock_read (id=0x960dc50, fd=13, events=1, ignore=0x0) at chan_sip.c:8598 #6 0x0805378f in ast_io_wait (ioc=0x960dc10, howlong=106) at io.c:267 #7 0xf6ec89b2 in do_monitor (data=0x0) at chan_sip.c:8745 #8 0x008661d5 in start_thread () from /lib/tls/libpthread.so.0 #9 0x007c02da in clone () from /lib/tls/libc.so.6 (gdb) If I can provide more information please let me know? Do you need a core dump file? Meanwhile I'll update glibc... Thanks in advance, JO Matthew Boehm wrote: Jose R. Ortiz Ubarri wrote: But if I load the module from boot or from the asterisk command load res_config_mysql.so then I get the Segmentation fault again. Where is your backtrace? I don't see a backtrace anywhere. Hi. My phone isn't working but I'm not going to let you see what I did to cause it to stop working. Send backtrace from the core file when it crashes. -Matthew ___ 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 -- Jose R. Ortiz Ubarri (CHEO), CS System Administrator / Programmer High Performance Computing facility - UPR Email: [EMAIL PROTECTED]|[EMAIL PROTECTED] Phone: 787-758-3054 Fax: 787-758-3058 ___ 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] Realtime Problem = Segmentation faults
#1 0xf6eb58c0 in build_peer (name=0x0, v=0x95fa370, realtime=0) at chan_sip.c:9255 #2 0xf6eb67b0 in find_peer (peer=0x0, sin=0x9642fd4, realtime=1) at chan_sip.c:1222 I just updated to newest CVS and visited those line numbers above. 9255 is in build_peer but find_peer is no where near 1222. Update to newest CVS. Recompile and try again. If it crashes just send what you sent that last time. That was good. -Matthew ___ 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] Realtime Problem = Segmentation faults
Jose R. Ortiz Ubarri wrote: (gdb) backtrace #0 0x007642b8 in strcasecmp () from /lib/tls/libc.so.6 #1 0xf6eb58c0 in build_peer (name=0x0, v=0x95fa370, realtime=0) at chan_sip.c:9255 #2 0xf6eb67b0 in find_peer (peer=0x0, sin=0x9642fd4, realtime=1) at chan_sip.c:1222 I do not see any realtime functions being called in there. find_peer seems to have found the peer in the sip.conf file. What CVS version (date stamp) are you running? The line numbers don't match what I have so it is hard to tell which strcasecmp failed. -Matthew ___ 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] Realtime Problem = Segmentation faults
Matthew: The .version file in the asterisk folder reads: CVS-HEAD-03/17/05-15:43:44 pd: I opened chan_sip.c at line 9255 and that line reads: peer = ASTOBJ_CONTAINER_FIND_UNLINK(peerl, name); No strcasecmp there... Thanks, JO Matthew Boehm wrote: Jose R. Ortiz Ubarri wrote: (gdb) backtrace #0 0x007642b8 in strcasecmp () from /lib/tls/libc.so.6 #1 0xf6eb58c0 in build_peer (name=0x0, v=0x95fa370, realtime=0) at chan_sip.c:9255 #2 0xf6eb67b0 in find_peer (peer=0x0, sin=0x9642fd4, realtime=1) at chan_sip.c:1222 I do not see any realtime functions being called in there. find_peer seems to have found the peer in the sip.conf file. What CVS version (date stamp) are you running? The line numbers don't match what I have so it is hard to tell which strcasecmp failed. -Matthew ___ 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 -- Jose R. Ortiz Ubarri (CHEO), CS System Administrator / Programmer High Performance Computing facility - UPR Email: [EMAIL PROTECTED]|[EMAIL PROTECTED] Phone: 787-758-3054 Fax: 787-758-3058 ___ 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] Realtime Problem = Segmentation faults
Matthew: I did the cvs checkout asterisk today. I think I have the latest version: The trace is: (gdb) backtrace #0 0x007642b8 in strcasecmp () from /lib/tls/libc.so.6 #1 0xf6eb58c0 in build_peer (name=0x0, v=0x92a42d0, realtime=0) at chan_sip.c:9255 #2 0xf6eb67b0 in find_peer (peer=0x0, sin=0x929bf4c, realtime=1) at chan_sip.c:1222 #3 0xf6ebea77 in check_user_full (p=0x929bdf0, req=0xf6e9bb50, cmd=0xf6e9bd64 SUBSCRIBE, uri=0xf6e9bd6e sip:[EMAIL PROTECTED]:5060, reliable=0, sin=0xf6e9bb40, ignore=0, mailbox=0xf6e920a0 , mailboxlen=0) at chan_sip.c:5844 #4 0xf6ec3129 in handle_request (p=0x929bdf0, req=0xf6e9bb50, sin=0xf6e9bb40, recount=0x0, nounlock=0xf6e9b9c8) at chan_sip.c:8384 #5 0xf6ec5281 in sipsock_read (id=0x927cc50, fd=13, events=1, ignore=0x0) at chan_sip.c:8598 #6 0x0805378f in ast_io_wait (ioc=0x927cc10, howlong=0) at io.c:267 #7 0xf6ec89b2 in do_monitor (data=0x0) at chan_sip.c:8745 #8 0x008661d5 in start_thread () from /lib/tls/libpthread.so.0 #9 0x007c02da in clone () from /lib/tls/libc.so.6 Thanks, JO Matthew Boehm wrote: #1 0xf6eb58c0 in build_peer (name=0x0, v=0x95fa370, realtime=0) at chan_sip.c:9255 #2 0xf6eb67b0 in find_peer (peer=0x0, sin=0x9642fd4, realtime=1) at chan_sip.c:1222 I just updated to newest CVS and visited those line numbers above. 9255 is in build_peer but find_peer is no where near 1222. Update to newest CVS. Recompile and try again. If it crashes just send what you sent that last time. That was good. -Matthew ___ 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 -- Jose R. Ortiz Ubarri (CHEO), CS System Administrator / Programmer High Performance Computing facility - UPR Email: [EMAIL PROTECTED]|[EMAIL PROTECTED] Phone: 787-758-3054 Fax: 787-758-3058 ___ 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] Realtime problem
Clay, Can you post your extconfig.conf and your database schema? If you want to load your static sip configuration into a database, follow these instructions: http://www.voip-info.org/tiki-index.php?page=Asterisk%20RealTime%20Static I haven't loaded a static file into the database using RealTime, but used this method and it works great: http://www.voip-info.org/tiki-index.php?page=Asterisk%20RealTime%20Sip I placed a timestamp column so I can track when the most recent successful registration occurred. -- -- Table structure for table `sip_buddies` -- CREATE TABLE `sip_buddies` ( `uniqueid` int(11) NOT NULL auto_increment, `name` varchar(30) NOT NULL default '', `accountcode` varchar(30) default NULL, `amaflags` char(1) default NULL, `callgroup` varchar(30) default NULL, `callerid` varchar(50) default NULL, `canreinvite` char(1) default NULL, `context` varchar(30) default NULL, `defaultip` varchar(15) default NULL, `dtmfmode` varchar(7) default NULL, `fromuser` varchar(50) default NULL, `fromdomain` varchar(31) default NULL, `host` varchar(31) NOT NULL default '', `incominglimit` char(2) default NULL, `outgoinglimit` char(2) default NULL, `insecure` char(1) default NULL, `language` char(2) default NULL, `mailbox` varchar(50) default NULL, `md5secret` varchar(32) default NULL, `nat` varchar(5) default NULL, `permit` varchar(95) default NULL, `deny` varchar(95) default NULL, `pickupgroup` varchar(10) default NULL, `port` varchar(5) NOT NULL default '', `qualify` varchar(4) default NULL, `restrictcid` char(1) default NULL, `rtptimeout` char(3) default NULL, `rtpholdtimeout` char(3) default NULL, `secret` varchar(30) default NULL, `type` varchar(6) NOT NULL default '', `username` varchar(30) NOT NULL default '', `allow` varchar(100) default NULL, `disallow` varchar(100) default NULL, `regseconds` int(11) NOT NULL default '0',; `ipaddr` varchar(15) NOT NULL default '', `ts` timestamp(14) NOT NULL, PRIMARY KEY (`uniqueid`), UNIQUE KEY `name` (`name`), KEY `name_2` (`name`) ) TYPE=MyISAM; -extconfig.conf- == ; Static and realtime external configuration ; engine configuration ; ; Please read doc/README.extconfig for basic table ; formatting information. ; [settings] ; ; Static configuration files: ; ; file.conf = driver,database[,table] ; ; maps a particular configuration file to the given ; database driver, database and table (or uses the ; name of the file as the table if not specified) ; ;uncomment to load queues.conf via the odbc engine. ; ;sip.conf = odbc,asterisk,sip ; ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; ;iaxfriends = odbc,asterisk sipfriends = mysql,asterisk,sip_buddies ;voicemail = odbc,asterisk -res_mysql.conf- ; ; Sample configuration for res_config_mysql.c ; ; The value of dbhost may be either a hostname or an IP address. ; If dbhost is commented out or the string localhost, a connection ; to the local host is assumed and dbsock is used instead of TCP/IP ; to connect to the server. ; [general] ;dbhost = 127.0.0.1 dbname = asterisk dbuser = [removed] dbpass = [removed] dbport = 3306 dbsock = /var/run/mysqld/mysqld.sock On Tuesday 14 December 2004 09:50 pm, Clay Reiche wrote: I'm having trouble with the Realtime setup. I've followed the instructions on voip-info using odbc but I get this message during asterisk boot: Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load config sip.conf, SIP disabled == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' And my device(s) won't register. I don't even see them attempt the registration...(from the CLI in ery verbose.) Maybe I'm not using the right version of asterisk??? Is that possible and how would I know? My show version gives me this: *CLI show version Asterisk CVS-v1-0-12/08/04-16:50:05 built by [EMAIL PROTECTED] on a i686 running Linux *CLI Any help would be appreciated. Thanks! Clay Reiche -- Brian Wilkins Software Engineer [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 321.308.4000 x33 http://www.hcc.net ___ 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] Realtime problem
I see that you are connecting to mysql locally. Take a look at this error: Dec 14 15:31:01 WARNING[8102]: MySQL database sock file not specified. Using default And then take a look at your res_mysql.conf. I see the error, do you? -Matthew - Original Message - From: Bruce Komito [EMAIL PROTECTED] To: Clay Reiche [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Tuesday, December 14, 2004 5:21 PM Subject: Re: [Asterisk-Users] Realtime problem I'm having exactly the same problem. I have sip.conf rows in the sql table (ast_config), and removed the /etc/asterisk/sip.conf file. Now I have no sip devices. It's as though realtime is not looking for the sip.conf rows in the table. This is my extconfig.conf: [settings] ; Static configuration files: ; file.conf = driver,database[,table] sip.conf = mysql,asteriskcdrdb,ast_config voicemail.conf = mysql,asteriskcdrdb,ast_config This is my res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = asteriskcdrdb dbuser = asterisk dbpass = none dbport = 3306 dbsock = =/var/lib/mysql/mysql.sock These are the startup messages I get when I start * (not voicemail.conf is loaded via mysql but not sip.conf: Dec 14 15:31:01 NOTICE[8102]: res_odbc loaded. Dec 14 15:31:01 NOTICE[8102]: Registered Config Engine odbc Dec 14 15:31:01 NOTICE[8102]: Registered Config Engine mysql Dec 14 15:31:01 NOTICE[8102]: Unable to load config sip.conf, SIP disabled Dec 14 15:31:01 WARNING[8102]: Unable to open IAX timing interface: No such device Dec 14 15:31:01 ERROR[8102]: Unable to load config iax.conf Dec 14 15:31:01 WARNING[8102]: Unable to get our IP address, Skinny disabled Dec 14 15:31:01 WARNING[8102]: Unable to open /dev/dsp: No such device Dec 14 15:31:01 WARNING[8102]: Requested contexts didn't get merged Dec 14 15:31:01 NOTICE[8102]: Loading Config voicemail.conf via mysql engine Dec 14 15:31:01 WARNING[8102]: MySQL database sock file not specified. Using default Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Tue, 14 Dec 2004, Clay Reiche wrote: I'm having trouble with the Realtime setup. I've followed the instructions on voip-info using odbc but I get this message during asterisk boot: Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load config sip.conf, SIP disabled == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' And my device(s) won't register. I don't even see them attempt the registration...(from the CLI in ery verbose.) Maybe I'm not using the right version of asterisk??? Is that possible and how would I know? My show version gives me this: *CLI show version Asterisk CVS-v1-0-12/08/04-16:50:05 built by [EMAIL PROTECTED] on a i686 running Linux *CLI Any help would be appreciated. Thanks! Clay Reiche This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-12-14%5C45f16737f297472c8726ed904c2e44c6C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Realtime problem
I'm having exactly the same problem. I have sip.conf rows in the sql table (ast_config), and removed the /etc/asterisk/sip.conf file. Now I have no sip devices. It's as though realtime is not looking for the sip.conf rows in the table. This is my extconfig.conf: [settings] ; Static configuration files: ; file.conf = driver,database[,table] sip.conf = mysql,asteriskcdrdb,ast_config voicemail.conf = mysql,asteriskcdrdb,ast_config This is my res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = asteriskcdrdb dbuser = asterisk dbpass = none dbport = 3306 dbsock = =/var/lib/mysql/mysql.sock These are the startup messages I get when I start * (not voicemail.conf is loaded via mysql but not sip.conf: Dec 14 15:31:01 NOTICE[8102]: res_odbc loaded. Dec 14 15:31:01 NOTICE[8102]: Registered Config Engine odbc Dec 14 15:31:01 NOTICE[8102]: Registered Config Engine mysql Dec 14 15:31:01 NOTICE[8102]: Unable to load config sip.conf, SIP disabled Dec 14 15:31:01 WARNING[8102]: Unable to open IAX timing interface: No such device Dec 14 15:31:01 ERROR[8102]: Unable to load config iax.conf Dec 14 15:31:01 WARNING[8102]: Unable to get our IP address, Skinny disabled Dec 14 15:31:01 WARNING[8102]: Unable to open /dev/dsp: No such device Dec 14 15:31:01 WARNING[8102]: Requested contexts didn't get merged Dec 14 15:31:01 NOTICE[8102]: Loading Config voicemail.conf via mysql engine Dec 14 15:31:01 WARNING[8102]: MySQL database sock file not specified. Using default Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Tue, 14 Dec 2004, Clay Reiche wrote: I'm having trouble with the Realtime setup. I've followed the instructions on voip-info using odbc but I get this message during asterisk boot: Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load config sip.conf, SIP disabled == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' And my device(s) won't register. I don't even see them attempt the registration...(from the CLI in ery verbose.) Maybe I'm not using the right version of asterisk??? Is that possible and how would I know? My show version gives me this: *CLI show version Asterisk CVS-v1-0-12/08/04-16:50:05 built by [EMAIL PROTECTED] on a i686 running Linux *CLI Any help would be appreciated. Thanks! Clay Reiche This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-12-14%5C45f16737f297472c8726ed904c2e44c6C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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
