Re: [asterisk-users] Fwd: Errors Compiling Libpri-1.4.13
2012/11/19 Shaun Ruffell sruff...@digium.com On Mon, Nov 19, 2012 at 08:47:23PM +0100, Adolphus Enaboifo wrote: .. I get errors while trying to compile Libpri 1.4.13. (check attachment} Can you guys please help me prescribe a fix. [snip] gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pridump.o -MF .pridump.o.d -MP -c -o pridump.o pridump.c pridump.c:45:24: fatal error: dahdi/user.h: No such file or directory compilation terminated. make: *** [pridump.o] Error 1 New in lipri 1.4.13 is a default dependency on DAHDI [1]. You should be good to go if you make sure that DAHDI is installed before compiling libpri. My installation scripts were designed to build things this way: libpri dahdi-linux dahdi-tools asterisk Is it now requested to build this way ? dahdi-linux libpri dahdi-tools asterisk [1] http://svnview.digium.com/svn/libpri?view=revisionrevision=2294 Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 not starting (illegal instruction core dumped)
Hi List members, Thanks for the support so far as I try to install and test my first asterisk system. I was able to finally install asterisk-1.8.18.0 with libpri-1.4.13 and dahdi-linux-complete-2.6.1+2.6.1 according to the instructions given in the online documentation (asterisk the definitive guide). But while trying to start asterisk with the following command /usr/sbin/asterisk -cvvv or /usr/sbin/asterisk -c I get the message Illegal instruction (core dumped) Kindly advice on what to do. thanks Adolphus Enaboifo -- _ -- 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 not starting (illegal instruction core dumped)
I suspect you have something wrong in your server hardware... have you tried running a memtest? Leandro 2012/11/27 Adolphus Enaboifo adolphus.enabo...@osenkorp.com Hi List members, Thanks for the support so far as I try to install and test my first asterisk system. I was able to finally install asterisk-1.8.18.0 with libpri-1.4.13 and dahdi-linux-complete-2.6.1+2.6.1 according to the instructions given in the online documentation (asterisk the definitive guide). But while trying to start asterisk with the following command /usr/sbin/asterisk -cvvv or /usr/sbin/asterisk -c I get the message Illegal instruction (core dumped) Kindly advice on what to do. thanks Adolphus Enaboifo -- _ -- 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 11 - Security event logging over syslog
Am 24.11.2012 um 11:25 schrieb Michael Keuter: Am 24.11.2012 um 01:33 schrieb Matthew Jordan: On 11/22/2012 04:00 PM, Michael Keuter wrote: Hi all, I am just testing with Asterisk 11.01. The SIP security event logging works fine for me for console and file logging, but the security events are not logged over syslog. logger.conf: ... syslog.local0 = notice,warning,error,security Is this on purpose, a fault on my side, or is this a bug? No, that should work. What's the output of 'logger show channels'? -- Matthew Jordan logger show channels Channel Type StatusConfiguration --- --- syslog.local0 Syslog Enabled- NOTICE WARNING ERROR SECURITY /var/log/asterisk/security_log File Enabled- SECURITY Console Enabled- NOTICE WARNING ERROR SECURITY Everything else except security works fine over syslog. Michael http://www.mksolutions.info I created an issue on the bugtracker for this: https://issues.asterisk.org/jira/browse/ASTERISK-20744 Michael http://www.mksolutions.info smime.p7s Description: S/MIME cryptographic signature -- _ -- 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] SIP Debugging Information..
Michael L. Young wrote: If I am reading this right, it looks like a BYE is coming in from the far end, Bandwidth.com. Prior to that, Asterisk retransmits the OK to Bandwidth.com's INVITE twice. It doesn't look like Bandwidth.com receives any of them, because they never respond with an ACK. Since, from Bandwidth.com's perspective, the call is never setup, they terminate it with a BYE. It could just be a NAT issue, but there are two things I really don't understand about the SIP dialog: 1) It starts with an ACK from Bandwidth.com. Is it possible that the debugging output is missing the beginning of the dialog? 2) Every timestamp is Nov 23 15:43:13. I don't think the SIP session timers on either end should be expiring quickly enough for this to happen. Do other calls originating from Bandwidth.com work properly? If so, comparing the SIP from a working call to a failed call may be revealing. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Queue logging
Hello, at the moment I am logging queues into a MySQL DB, but this can quickly become a lot of information. Is there a way to exclude certain queues from being logged into the queue log ? Thanks, Jonas. -- _ -- 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] Queue logging
Are you using triggering? If so, perhaps you could modify the trigger values. PS asterisk version? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, November 27, 2012 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Queue logging Hello, at the moment I am logging queues into a MySQL DB, but this can quickly become a lot of information. Is there a way to exclude certain queues from being logged into the queue log ? Thanks, Jonas. -- _ -- 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] Fwd: Errors Compiling Libpri-1.4.13
On Mon, Nov 19, 2012 at 08:47:23PM +0100, Adolphus Enaboifo wrote: .. I get errors while trying to compile Libpri 1.4.13. (check attachment} Can you guys please help me prescribe a fix. [snip] gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pridump.o -MF .pridump.o.d -MP -c -o pridump.o pridump.c pridump.c:45:24: fatal error: dahdi/user.h: No such file or directory compilation terminated. make: *** [pridump.o] Error 1 New in lipri 1.4.13 is a default dependency on DAHDI [1]. You should be good to go if you make sure that DAHDI is installed before compiling libpri. My installation scripts were designed to build things this way: libpri dahdi-linux dahdi-tools asterisk Is it now requested to build this way ? dahdi-linux libpri dahdi-tools asterisk Yes. Previously, the libpri utilities were not built by default. Now they are. It is the utilities that have the dependency on DAHDI. It makes more sense to build from the ground up anyway. dahdi-linux \__ Hardware level drivers and utilities dahdi-tools / libpri - Layer 2/Layer 3 protocols asterisk Richard -- _ -- 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] SIP password probe
It's an open source project. Pay a programmer or make the modification yourself and submit a patch. On Sat, Nov 24, 2012 at 4:51 PM, Ron Wheeler rwhee...@artifact-software.com wrote: I looking through my logs, I found that people where probing my SIP accounts looking for passwords. Asterisk was helping them out by processing hundreds of requests per minute. I did a bit of Googling and this seems to be a frequent knock against Asterisk's security. It would seem pretty simple to add a configuration setting to sip.conf to delay the response to a bad account or password. There is a half measure to confuse the probe by sending the same error return for either error. It appears that many people have complained that this should be the default setting only changed if your are debugging a problem. There is no reason for a working system to ever have bad passwords so this is clearly an attack in almost every case. A simple delay would solve the problem for most people who use reasonable passwords. I had to install fail2ban which is a PITA but thanks to someone's clear recipe, I was able to get it working. I hope that this can be worked into a release soon. Ron -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] 23B + D Asterisk TDMoE (packetized PRI) link question.
Can anyone help me find a specification for the 23B + D Asterisk TDMoE (packetized PRI) link? I am having occasional packets arrive from the Asterisk Server that do not have the standard four leading characters in front of the MAC addresses, and am not sure what this implies. For now, I discard such packets. Also, I'm getting HDLC abort messages from the Server that occur almost exclusively in heavy call set-up traffic, despite that the PRI-style messages in both directions appear to have correct indices, Call Reference Numbers, etc. If you can answer the question or if you know where to go to get the answer that information would be appreciated. Thank you, Aaron Bailey -- _ -- 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] Queue logging
Hello, I am not using triggering (what is this ?). Just using extconfig.conf Asterisk 1.8.12.2 Kind regards, Jonas. On 27-11-12 17:28, Danny Nicholas wrote: Are you using triggering? If so, perhaps you could modify the trigger values. PS asterisk version? *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Tuesday, November 27, 2012 10:21 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Queue logging Hello, at the moment I am logging queues into a MySQL DB, but this can quickly become a lot of information. Is there a way to exclude certain queues from being logged into the queue log ? Thanks, Jonas. -- _ -- 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
[asterisk-users] AEL syntax consistency
How consistent has the syntax for extensions.ael been from version to version? extensions.conf has annoyed me in this regard. i.e.: commas to pipes, pipes back to commas, macro to gosub, etc etc. -- _ -- 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 not starting (illegal instruction core dumped)
Build for the wrong processor type? Wrong arch? Kernel binary format support? On Tue, Nov 27, 2012 at 6:05 AM, Adolphus Enaboifo adolphus.enabo...@osenkorp.com wrote: Hi List members, Thanks for the support so far as I try to install and test my first asterisk system. I was able to finally install asterisk-1.8.18.0 with libpri-1.4.13 and dahdi-linux-complete-2.6.1+2.6.1 according to the instructions given in the online documentation (asterisk the definitive guide). But while trying to start asterisk with the following command /usr/sbin/asterisk -cvvv or /usr/sbin/asterisk -c I get the message Illegal instruction (core dumped) Kindly advice on what to do. thanks Adolphus Enaboifo -- _ -- 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 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] SIP password probe
On 27/11/2012 12:58 PM, Christopher Harrington wrote: It's an open source project. Pay a programmer or make the modification yourself and submit a patch. You don't really want me coding! I have solved the problem for me. Just add it to the queue of enhancements for the next time someone is working on SIP. Ron On Sat, Nov 24, 2012 at 4:51 PM, Ron Wheeler rwhee...@artifact-software.com mailto:rwhee...@artifact-software.com wrote: I looking through my logs, I found that people where probing my SIP accounts looking for passwords. Asterisk was helping them out by processing hundreds of requests per minute. I did a bit of Googling and this seems to be a frequent knock against Asterisk's security. It would seem pretty simple to add a configuration setting to sip.conf to delay the response to a bad account or password. There is a half measure to confuse the probe by sending the same error return for either error. It appears that many people have complained that this should be the default setting only changed if your are debugging a problem. There is no reason for a working system to ever have bad passwords so this is clearly an attack in almost every case. A simple delay would solve the problem for most people who use reasonable passwords. I had to install fail2ban which is a PITA but thanks to someone's clear recipe, I was able to get it working. I hope that this can be worked into a release soon. Ron -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com mailto:rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 tel:866-970-2435%2C%20ext%20102 -- _ -- 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 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] SIP password probe
You might want to share the know how over here if its not a chan_sip patch. Mitul On Nov 28, 2012 12:28 AM, Ron Wheeler rwhee...@artifact-software.com wrote: On 27/11/2012 12:58 PM, Christopher Harrington wrote: It's an open source project. Pay a programmer or make the modification yourself and submit a patch. You don't really want me coding! I have solved the problem for me. Just add it to the queue of enhancements for the next time someone is working on SIP. Ron On Sat, Nov 24, 2012 at 4:51 PM, Ron Wheeler rwhee...@artifact-software.com wrote: I looking through my logs, I found that people where probing my SIP accounts looking for passwords. Asterisk was helping them out by processing hundreds of requests per minute. I did a bit of Googling and this seems to be a frequent knock against Asterisk's security. It would seem pretty simple to add a configuration setting to sip.conf to delay the response to a bad account or password. There is a half measure to confuse the probe by sending the same error return for either error. It appears that many people have complained that this should be the default setting only changed if your are debugging a problem. There is no reason for a working system to ever have bad passwords so this is clearly an attack in almost every case. A simple delay would solve the problem for most people who use reasonable passwords. I had to install fail2ban which is a PITA but thanks to someone's clear recipe, I was able to get it working. I hope that this can be worked into a release soon. Ron -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] SIP password probe
I had to install fail2ban and configure it to watch Asterisk. Ron On 27/11/2012 2:11 PM, Mitul Limbani wrote: You might want to share the know how over here if its not a chan_sip patch. Mitul On Nov 28, 2012 12:28 AM, Ron Wheeler rwhee...@artifact-software.com mailto:rwhee...@artifact-software.com wrote: On 27/11/2012 12:58 PM, Christopher Harrington wrote: It's an open source project. Pay a programmer or make the modification yourself and submit a patch. You don't really want me coding! I have solved the problem for me. Just add it to the queue of enhancements for the next time someone is working on SIP. Ron On Sat, Nov 24, 2012 at 4:51 PM, Ron Wheeler rwhee...@artifact-software.com mailto:rwhee...@artifact-software.com wrote: I looking through my logs, I found that people where probing my SIP accounts looking for passwords. Asterisk was helping them out by processing hundreds of requests per minute. I did a bit of Googling and this seems to be a frequent knock against Asterisk's security. It would seem pretty simple to add a configuration setting to sip.conf to delay the response to a bad account or password. There is a half measure to confuse the probe by sending the same error return for either error. It appears that many people have complained that this should be the default setting only changed if your are debugging a problem. There is no reason for a working system to ever have bad passwords so this is clearly an attack in almost every case. A simple delay would solve the problem for most people who use reasonable passwords. I had to install fail2ban which is a PITA but thanks to someone's clear recipe, I was able to get it working. I hope that this can be worked into a release soon. Ron -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com mailto:rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 tel:866-970-2435%2C%20ext%20102 -- _ -- 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 -- -Chris Harrington ACSDi Office: 763.559.5800 tel:763.559.5800 Mobile Phone: 612.326.4248 tel:612.326.4248 -- Ron Wheeler President Artifact Software Inc email:rwhee...@artifact-software.com mailto:rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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 -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Queue logging
Triggering is a MYSQL mechanism that forces database action on specified conditions. My best guess is that you would have to tweak addons/res_config_mysql.c to be able to filter logs. It would probably be easier to write a daemon to clear the unwanted data on a periodic basis. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, November 27, 2012 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue logging Hello, I am not using triggering (what is this ?). Just using extconfig.conf Asterisk 1.8.12.2 Kind regards, Jonas. On 27-11-12 17:28, Danny Nicholas wrote: Are you using triggering? If so, perhaps you could modify the trigger values. PS asterisk version? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, November 27, 2012 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Queue logging Hello, at the moment I am logging queues into a MySQL DB, but this can quickly become a lot of information. Is there a way to exclude certain queues from being logged into the queue log ? Thanks, Jonas. -- _ -- 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
[asterisk-users] SIP not answering on one trunk.
I have 2 analog trunks.
They answer the incoming call, do the welcome message, ask for the
extension, when a valid extension is entered it rings the right SIP
phone BUT
when the SIP phone is answered, the SIP phone keeps ringing and the call
is not connected.
If the phone is not answered it goes to voicemail correctly.
[DID_866nnn] IAX that works
include = DID_866nnn_default
[DID_866nnn_default]
exten = 866n,1,Goto(voicemenu-artifact-en,s,1)
[DID_trunk_1]
include = DID_trunk_1_default
[DID_trunk_1_default]
exten = s,1,Goto(voicemenu-artifact-fr,s,1)
[DID_trunk_2]
include = DID_trunk_2_default
[DID_trunk_2_default]
exten = s,1,Goto(voicemenu-home,s,1)
[voicemenu-artifact-en]
;ArtifactEnglishFirst
include = default
include = conferences
exten = s,1,Answer
exten = s,n,Set(CALLERID(name)=Art-${CALLERID(name)})
exten = s,n,Wait(0.5)
exten = s,n,Background(record/HelloArtifactEnglish)
exten = s,n(menu),Background(record/DialExtensionEnglish)
exten = s,n,WaitExten(3)
exten = 0,1,Goto(inbound-reception,s,1)
exten = 9,1,Goto(changeLanguageFrArtifact,s,1)
exten = #,1,Directory(default,default,f)
exten = t,1,Goto(inbound-reception,s,1)
exten = i,1,Goto(voicemenu-artifact-en,s,menu)
My IAX trunks work.
Log of dialing in on Trunk2 - answering SIP 102 and waiting while it
continued to ring.
[2012-11-27 14:43:52] VERBOSE[3589] sig_analog.c: -- Starting simple
switch on 'DAHDI/2-1'
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: -- Executing
[s@DID_trunk_2:1] Goto(DAHDI/2-1, voicemenu-home,s,1) in new stack
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: -- Goto (voicemenu-home,s,1)
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: -- Executing
[s@voicemenu-home:1] Answer(DAHDI/2-1, ) in new stack
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: -- Executing
[s@voicemenu-home:2] Set(DAHDI/2-1, CALLERID(name)=Home-ARTIFACT
LOGICI) in new stack
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: -- Executing
[s@voicemenu-home:3] Wait(DAHDI/2-1, 0.5) in new stack
[2012-11-27 14:43:53] VERBOSE[3589] pbx.c: -- Executing
[s@voicemenu-home:4] BackGround(DAHDI/2-1,
record/HelloAnnetteAndRon) in new stack
[2012-11-27 14:43:53] VERBOSE[3589] file.c: -- DAHDI/2-1 Playing
'record/HelloAnnetteAndRon.ulaw' (language 'en')
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: == CDR updated on DAHDI/2-1
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: -- Executing
[102@voicemenu-home:1] Macro(DAHDI/2-1, stdexten,102,SIP/102) in new
stack
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: -- Executing
[s@macro-stdexten:1] Set(DAHDI/2-1, __DYNAMIC_FEATURES=) in new stack
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: -- Executing
[s@macro-stdexten:2] Set(DAHDI/2-1, ORIG_ARG1=102) in new stack
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: -- Executing
[s@macro-stdexten:3] GotoIf(DAHDI/2-1, 0?6:4) in new stack
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: -- Goto (macro-stdexten,s,4)
[2012-11-27 14:43:56] VERBOSE[3589] pbx.c: -- Executing
[s@macro-stdexten:4] Dial(DAHDI/2-1, SIP/102,20,) in new stack
[2012-11-27 14:43:56] VERBOSE[3589] netsock2.c: == Using SIP RTP CoS
mark 5
[2012-11-27 14:43:56] VERBOSE[3589] app_dial.c: -- Called SIP/102
[2012-11-27 14:43:56] VERBOSE[3589] app_dial.c: -- SIP/102-0002
is ringing
[2012-11-27 14:44:01] VERBOSE[3589] app_dial.c: -- SIP/102-0002
answered DAHDI/2-1
Rang for a whole minute and a half until I hung up the DAHDI/2-1
[2012-11-27 14:45:42] VERBOSE[3589] pbx.c: -- Executing
[h@voicemenu-home:1] Hangup(DAHDI/2-1, ) in new stack
[2012-11-27 14:45:42] VERBOSE[3589] features.c: == Spawn extension
(voicemenu-home, h, 1) exited non-zero on 'DAHDI/2-1'
[2012-11-27 14:45:42] VERBOSE[3589] app_macro.c: == Spawn extension
(macro-stdexten, s, 4) exited non-zero on 'DAHDI/2-1' in macro 'stdexten'
[2012-11-27 14:45:42] VERBOSE[3589] pbx.c: == Spawn extension
(voicemenu-home, 102, 1) exited non-zero on 'DAHDI/2-1'
[2012-11-27 14:45:42] VERBOSE[3589] sig_analog.c: -- Hanging up on
'DAHDI/2-1'
[2012-11-27 14:45:42] VERBOSE[3589] chan_dahdi.c: -- Hungup 'DAHDI/2-1'
(END)
--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102
--
_
-- 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] SDK for Asterisk, where is it?
Hello; I remember that I saw at asterisk website (this was maybe before 1 year or around) some pages are talking about having SDK and APIs for asterisk that will be used to build softphone for mobile and will be used to build some applications for asterisk, also it was mentioned in this page that this project is resuming after it was stopped before. But now I am not able to find those pages about this subject any more. What happened about it? Are they going to really do it or it is cancelled. If it is still existed, where is the link for this and what it is situation? Regards Bilal -- _ -- 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] Queue logging
Ah OK, that triggering I know. I though maybe there was some kind of setting on a per queue base that could control the logging, like there is amaflags on a peer. Jonas. On 27-11-12 20:53, Danny Nicholas wrote: Triggering is a MYSQL mechanism that forces database action on specified conditions. My best guess is that you would have to tweak addons/res_config_mysql.c to be able to filter logs. It would probably be easier to write a daemon to clear the unwanted data on a periodic basis. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Tuesday, November 27, 2012 12:27 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Queue logging Hello, I am not using triggering (what is this ?). Just using extconfig.conf Asterisk 1.8.12.2 Kind regards, Jonas. On 27-11-12 17:28, Danny Nicholas wrote: Are you using triggering? If so, perhaps you could modify the trigger values. PS asterisk version? *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Tuesday, November 27, 2012 10:21 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Queue logging Hello, at the moment I am logging queues into a MySQL DB, but this can quickly become a lot of information. Is there a way to exclude certain queues from being logged into the queue log ? Thanks, Jonas. -- _ -- Bandwidth and Colocation Provided byhttp://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
