Re: [asterisk-users] Crash asterisk res_odbc
I am sure it is centos 6.7. That was we what we decided together to have the Colo to use Sent from my Verizon Wireless 4G LTE smartphone Original message From: Leandro Dardini Date: 2/28/2016 3:49 PM (GMT-05:00) To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Crash asterisk res_odbc Which operating system are you using? I have experienced the same problem on several OS except for CentOS 6. I suppose an ODBC problem on newer OS version. Leandro Il 24/Feb/2016 05:30 PM, "Maxime" ha scritto: Dear list, i have a issue Asterisk crash (Module res_odbc exactly) after the same log who is "ERROR[23805] astobj2.c: bad magic number..." you will see on the log : Today [2016-02-24 16:00:38] ERROR[23805] astobj2.c: bad magic number 0x552f302e for 0x7fe3505b3958 [2016-02-24 16:00:44] Asterisk 11.2-cert1 built by root @ Voice_server on a x86_64 running Linux on 2013-04-09 14:16:57 UTC [2016-02-24 16:00:44] NOTICE[31321] loader.c: 2 modules will be loaded. [2016-02-24 16:00:44] NOTICE[31321] res_odbc.c: Connecting asterisk [2016-02-24 16:00:44] NOTICE[31321] res_odbc.c: res_odbc: Connected to asterisk [MySQL-asterisk] [2016-02-24 16:00:44] NOTICE[31321] res_odbc.c: Registered ODBC class 'asterisk' dsn->[MySQL-asterisk] Yesterday : [2016-02-23 15:59:12] ERROR[19824] astobj2.c: bad magic number 0x20 for 0x27a5558 [2016-02-23 15:59:18] Asterisk 11.2-cert1 built by root @ Voice_server on a x86_64 running Linux on 2013-04-09 14:16:57 UTC [2016-02-23 15:59:18] NOTICE[23791] loader.c: 2 modules will be loaded. [2016-02-23 15:59:18] NOTICE[23791] res_odbc.c: Connecting asterisk [2016-02-23 15:59:18] NOTICE[23791] res_odbc.c: res_odbc: Connected to asterisk [MySQL-asterisk] [2016-02-23 15:59:18] NOTICE[23791] res_odbc.c: Registered ODBC class 'asterisk' dsn->[MySQL-asterisk] Effect : many trunk sip are down during few minutes Oddness : same hours On google i found many times "memory corruption was the assumption" ... Have you ever seen this kind of problem ? thank you in advance Version : Asterisk 11.2-cert1 Os : Debian 7-64 -- Maxime -- _ -- 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] Crash asterisk res_odbc
Which operating system are you using? I have experienced the same problem on several OS except for CentOS 6. I suppose an ODBC problem on newer OS version. Leandro Il 24/Feb/2016 05:30 PM, "Maxime" ha scritto: > Dear list, > > i have a issue > > Asterisk crash (Module res_odbc exactly) after the same log who is > "*ERROR[23805] > astobj2.c: bad magic number...*" > you will see on the log : > > Today > > [2016-02-24 16:00:38] ERROR[23805] *astobj2.c: bad magic number > 0x552f302e for 0x7fe3505b3958* > [2016-02-24 16:00:44] Asterisk 11.2-cert1 built by root @ Voice_server on > a x86_64 running Linux on 2013-04-09 14:16:57 UTC > [2016-02-24 16:00:44] NOTICE[31321] loader.c: 2 modules will be loaded. > [2016-02-24 16:00:44] NOTICE[31321] res_odbc.c: Connecting asterisk > [2016-02-24 16:00:44] NOTICE[31321] res_odbc.c: res_odbc: Connected to > asterisk [MySQL-asterisk] > [2016-02-24 16:00:44] NOTICE[31321] res_odbc.c: Registered ODBC class > 'asterisk' dsn->[MySQL-asterisk] > > Yesterday : > > [2016-02-23 15:59:12] ERROR[19824] *astobj2.c: bad magic number 0x20 for > 0x27a5558* > [2016-02-23 15:59:18] Asterisk 11.2-cert1 built by root @ Voice_server on > a x86_64 running Linux on 2013-04-09 14:16:57 UTC > [2016-02-23 15:59:18] NOTICE[23791] loader.c: 2 modules will be loaded. > [2016-02-23 15:59:18] NOTICE[23791] res_odbc.c: Connecting asterisk > [2016-02-23 15:59:18] NOTICE[23791] res_odbc.c: res_odbc: Connected to > asterisk [MySQL-asterisk] > [2016-02-23 15:59:18] NOTICE[23791] res_odbc.c: Registered ODBC class > 'asterisk' dsn->[MySQL-asterisk] > > Effect : many trunk sip are down during few minutes > Oddness : same hours > > On google i found many times "memory corruption was the assumption" ... > > Have you ever seen this kind of problem ? > > thank you in advance > > Version : Asterisk 11.2-cert1 > Os : Debian 7-64 > > -- > > Maxime > > -- > _ > -- 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] Handle a call if one phone of a ring, group is busy
I do it via a group count: main call handling: exten => sub123,n,Set(GROUP()=11122345) ... the main routine calls subroutine: exten => general,1,GotoIf($["${busyonbusy}"="YES"]?100:200) exten => general,100,GotoIf($[ ${GROUP_COUNT()} > 1 ]?110:200) exten => general,110,Hangup(17) ; fehlercode 17 = SIPcode 486=user busy here exten => general,200,Return() ... that works as well and you can specify how many calls are allowed. regards, andre -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Handle a call if one phone of a ring group is busy
On Sun, 2016-02-28 at 01:43 +0100, Frank wrote: > Question: How to give a "busy signal" back to the caller if one > extension of a ring group is in use? Or redirect the call to voice mail? Found a solution! :-) exten => 7654321,1,GotoIf($["${DEVICE_STATE(SIP/111)}"="INUSE"]?Busy,1) exten => 7654321,n,GotoIf($["${DEVICE_STATE(SIP/222)}"="INUSE"]?Busy,1) exten => 7654321,n,GotoIf($["${DEVICE_STATE(SIP/333)}"="INUSE"]?Busy,1) exten => 7654321,n,Dial(SIP/111&SIP/222&SIP/333) exten => Busy,1,BUSY(10) exten => Busy,n,Hangup -- _ -- 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] 1000 analogue lines with asterisk
On Fri, Feb 19, 2016 at 10:52:34AM +1300, Daniel Harper wrote: > What about leaving the old PBX in place and trunking it via ISDN to the > asterisk server. > > We use rhino 24 channel bank but are 2U for rhino + 1U for patch panel. > (RJ21 cable so might be able to use existing ones if they are RJ21) > > Used USB xorcoms a while back, things may of changed but if one is down and > reboot server then asterisk doesn't come up. Not with recent versions of DAHDI. Assuming you set auto_assign_spans=0, DAHDI channel and span numbers are mapped in /etc/dahdi/assigned-spans.conf the order of their startup is irrelevant: the identification of channels attached to either serial number or connector, and not to semi-random initialization order. And this is why Asterisk could afford itself not to fail if DAHDI channels are missing. Originally it failed because missing devices meant that a device may be missing and thus existing channels may be misplaced. With a recent enough version of Asterisk, and with ignore_failed_channels = true in chan_dahdi.conf (the default as of 13. Makes sense also if you only have a single DAHDI device to begin with) Asterisk will not fail if channels are missing: they wil just come later. And indeed Asterisk can now allow DAHDI channels to be created after initialization, and thus Asterisk does not need to wait for all DAHDI devices to initialize before starting. (This is not specific to Xorcom hardware and should generally apply to any DAHDI hardware. I'm not exactly sure how this interacts with dynamic spans and interested to hear reports, probably in a different thread) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.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] app_swift crash asterisk 11.20.0-rc1
Not sure if I can provide much help, however I have had problems with Swift segmentation faults in the past. The problems I had were related to the license server causing an Asterisk segmentation fault any time security scans hit the server. That was with Asterisk 11.8. Cepstral has been unable to fix this issue and I'm considering using Nuance in the future. They did provide a workaround but that created an object leak, which forces a nightly restart of Asterisk. Program terminated with signal 11, Segmentation fault. #0 0x7fb08dd02061 in LM_set_port_number () from /opt/swift/lib/libswift.so.6 #0 0x7fb08dd02061 in LM_set_port_number () from /opt/swift/lib/libswift.so.6 I'd be interested to see if anyone has any ideas on your issue though. Bryan Burroughs On 02/27/2016 09:47 PM, Jeremy Kister wrote: I found the app_swift module (that I've been helping maintain) makes asterisk crash on versions higher than 11.19.0 - something that happened on 11.20.0-rc1 makes asterisk segfault. I realize app_swift is not a 'supported' module -- I'm just having a hard time finding the cause and am wondering if I could borrow anyone's eyes. of note, app_swift doesnt /always/ crash asterisk, e.g., when I call into asterisk from a phone and swift is in the dialplan, all seems fine. it seems that it's just when I make a callfile that dials out. a backtrace is at http://pastebin.com/Dfd4P8sK replication is easy (if you have swift): echo "testing 1 2 3" > /var/lib/asterisk/tts cat <<__EOE__ >> /etc/asterisk/extensions.conf [intercom] exten => _2XZ,1,SIPAddHeader(Alert-Info: Ring Answer) exten => _2XZ,n,Page(SIP/${EXTEN},diqA(local/intercom)) [tts] exten => s,1,Wait(1) exten => s,n,GotoIf($[0${LEN(${TEXT})} > 1]?text) exten => s,n,Set(SPEECH=${SHELL(cat /var/lib/asterisk/tts)}) exten => s,n,Goto(swift) exten => s,n(text),Set(SPEECH=${TEXT}) exten => s,n,NoOp(${SPEECH}) exten => s,n(swift),Swift(${SPEECH}) exten => s,n,Hangup __EOE__ cat <<__EOS__ > /var/spool/asterisk/tmp/test123 Channel: Local/221@intercom Callerid: "TTS" <0> MaxRetries: 2 WaitTime: 45 Context: tts Extension: s Priority: 1 __EOS__ mv /var/spool/asterisk/tmp/test123 /var/spool/asterisk/outgoing/test123 -- _ -- 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] Error compiling dahdi on CentOS 7
On Wed, Feb 24, 2016 at 03:55:09PM -0600, Carlos Chavez wrote: > I am having a problem trying to compile > dahdi-linux-complete-2.11.0+2.11.0 on a CentOS 7.2 server. Version 2.10.2 > compiles fine. Is there a new dependency for 2.11.0 that was not required > for previous versions? Here are some of the errors I get: > > INSTALL > /usr/src/dahdi-linux-complete-2.11.0+2.11.0/linux/drivers/dahdi/dahdi.ko > Can't read private key > INSTALL > /usr/src/dahdi-linux-complete-2.11.0+2.11.0/linux/drivers/dahdi/dahdi_dynamic.ko > Can't read private key > INSTALL > /usr/src/dahdi-linux-complete-2.11.0+2.11.0/linux/drivers/dahdi/dahdi_dynamic_eth.ko > Can't read private key I'm not sure what this is. However, > > /usr/bin/install: cannot stat ‘./dahdi_registration.8’: No such file or > directory > /usr/bin/install: cannot stat ‘./xpp_sync.8’: No such file or directory > /usr/bin/install: cannot stat ‘./lsdahdi.8’: No such file or directory > /usr/bin/install: cannot stat ‘./xpp_blink.8’: No such file or directory > /usr/bin/install: cannot stat ‘./dahdi_genconf.8’: No such file or directory > /usr/bin/install: cannot stat ‘./dahdi_hardware.8’: No such file or > directory > /usr/bin/install: cannot stat ‘./twinstar.8’: No such file or directory This one shouldl be fixed in dahdi-tools 2.11.1-rc1 . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.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