similar to: cdr_custom.conf in V12

Hi! I have set usegmtime=yes in cdr.conf, but unfortunately this is only for cdr-csv, not for cdr-custom. AFAIS there is no such option for cdr_custom.conf. Is there any workaround to get GMT timestamps in cdr-custom too? thanks klaus

I have a need to use cdr_custom and would like to know if anyone has gotten it to work with a mysql cdr backend, and any examples if possible -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050922/bca01301/attachment.htm

I was quite surprised to find a message in my in box from SellVOIP a day or two ago. It indicated I was running out of credit which was a surprise as I thought they'd gone under a large number of months back. So I ran upstairs, added their entry back to sip.conf, uncommented a couple of lines in extensions.conf and I'm again using sellvoip to make outgoing calls. The reason I was

On Thu, Oct 16, 2014 at 02:10:29PM -0400, Waiman Long wrote: > v11->v12: > - Based on PeterZ's version of the qspinlock patch > (https://lkml.org/lkml/2014/6/15/63). > - Incorporated many of the review comments from Konrad Wilk and > Paolo Bonzini. > - The pvqspinlock code is largely from my previous version with > PeterZ's way of going from queue tail

Hello Asterisk-users, [2013-08-18 05:56:29] NOTICE[17089][C-000000a8] chan_sip.c: Failed to authenticate device 390<sip:390 at xx.xx.xxx.xxx>;tag=2762c06e [2013-08-18 05:56:34] NOTICE[17089][C-000000a9] chan_sip.c: Failed to authenticate device 390<sip:390 at xx.xx.xxx.xxx>;tag=7b909220 I keep getting messages like this where the IP, xx.xx.xxx.xxx, is my own IP. How do I figure

Hello Asterisk, Would someone be kind enough as to add the issue: grp_lock error when compiling against pjproject and solution: delete the rogue install in /usr/local/include To the WIKI page about installing pjsip. I tried to update the WIKI but don't seem to have a way to do it. I know it's not supposed to happen and I know what I did wrong, but it's hard to imagine

Hello Asterisk, I've been running a CENTOS 5 box with Asterisk 14 and am trying to move to Asterisk 17.2 on a new Fedora Server 31 box. I built Asterisk from Source as I've always done and copied all the configuration files and other stuff from the old box. Everything comes up as expected and it all seems to work except I have one way audio. I'm still using SIP, not pjsip. As soon as

I get this when my brother in law tries to call in from his box to mine. WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have <100>, digest has <s> or after changing the register line: WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have <100>, digest has <199> I have done everything I can think of and still failure. Currently the

Hi all, The goal of the series is to give a chance to the architecture to validate VIRTIO device features. I changed VIRTIO_F_IOMMU_PLATFORM to VIRTIO_F_ACCESS_PLATFORM I forgot in drivers/virtio/Kconfig, and put back the inclusion of virtio_config.h for the definition of the callback in arch/s390/mm/init.c I wrongly removed in the last series. Regards, Pierre Pierre Morel (2): virtio: let

Hi all, The goal of the series is to give a chance to the architecture to validate VIRTIO device features. I changed VIRTIO_F_IOMMU_PLATFORM to VIRTIO_F_ACCESS_PLATFORM I forgot in drivers/virtio/Kconfig, and put back the inclusion of virtio_config.h for the definition of the callback in arch/s390/mm/init.c I wrongly removed in the last series. Regards, Pierre Pierre Morel (2): virtio: let

Yesterday I upgraded my Zaptel to 1.2.26 or I think that was it, the latest 1.2 version at downloads.digium.com. I have a Digium 4 card populated with 4 red FXO cards using channels 1,2 and 4. Channel 3 is not used. It's been working fine for a few years. After upgrading to 1.2.26 calls stopped coming in on channel 1, Channel 2 still worked fine and I could get dialtone and make calls

Hello all, I'm trying to create a 2x2 matrix, 32 times after unlist() so that I can convert the list to matrix. I've looked through the R archive but couldn't find the answer. There is what I've done. > f <- system("ls *.txt", intern=TRUE) > x <- lapply(f, read.table) > x [[1]] V1 V2 1 -27.3 14.4 2 29.0 -38.1 [[2]] V1 V2 1 14.4

Hello, I've installed a new Asterisk 17.0.0 on a Debian Buster system. This Asterisk instance is run by asterisk user (and group). I've got: # ls -l /etc/asterisk total 68 -rw-r--r-- 1 asterisk asterisk 501 nov. 18 19:12 asterisk.conf -rw-r--r-- 1 asterisk asterisk 135 nov. 18 18:57 cdr.conf -rw-r--r-- 1 asterisk asterisk 684 nov. 18 18:57 cdr_custom.conf -rw-r--r-- 1 asterisk

Hello Ira, Great! You are right! Through Chrome browser on my win7 and win8.1 got size 90MB tarball, through wget on my ubuntu got size 19MB gzipped file, thank you Ira and Karolin, great samba work! [~] # wget https://download.samba.org/pub/samba/stable/samba-4.1.19.tar.gz --no-check-certificate ... [~] # file samba-4.1.19.tar.gz samba-4.1.19.tar.gz: gzip compressed data, was

The Asterisk Development Team would like to announce the release of Asterisk 15.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.1.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

I show the file size as 19M on the FTP site. 95MB is the gunzipped size. I suspect something unzipped it as it downloaded, I've seen browsers do that. I also pulled the file personally, and found the sizes lined up: ira at ira-t430:~/Downloads [/dev/pts/1](64/0)$ ls -la samba-4.1.19.tar.gz -rw-rw-r--. 1 ira ira 19558250 Jun 23 20:16 samba-4.1.19.tar.gz ira at ira-t430:~/Downloads

The Asterisk Development Team would like to announce the release of Asterisk 14.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Hi, my Asterisk 1.2.9.1 suddenly freezed ("stop now" did not work!!) . I found the following errors inside /var/log/asterisk/message: Feb 14 14:55:41 ERROR[11273] rtp.c: Unable to allocate socket: Too many open files Feb 14 14:55:41 WARNING[11273] chan_sip.c: Unable to create RTP audio session: Too many open files Feb 14 14:55:43 WARNING[11273] acl.c: Cannot create socket Feb 14

The Asterisk Development Team would like to announce the release of Asterisk 13.18.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.18.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Dear all, I am having an issue with CDR logging. What I want to do is log jitter variable from RTPAUDIOQOS module into Master.csv at the end of each call. I am using asterisk version 1.4.26. For CDR purposes, I am using cdr_custom, and the content of my cdr_custom.conf is the following: [mappings] Master.csv =>