similar to: Xapian 1.2.13 released

I'm trying to compile latest zaptel-1.2.13 and I'm getting following errors: /usr/src/Asterisk-1.2.14/zaptel-1.2.13/xpp/xbus-core.c: In function ?debugfs_open?: /usr/src/Asterisk-1.2.14/zaptel-1.2.13/xpp/xbus-core.c:171: error: ?struct inode? has no member named ?i_private? make[5]: *** [/usr/src/Asterisk-1.2.14/zaptel-1.2.13/xpp/xbus-core.o] Error 1 make[4]: ***

Puppet Dashboard 1.2.13-rc1 is a maintenance and bugfix release candidate of Puppet Dashboard. This release is available for download at: https://downloads.puppetlabs.com/dashboard/puppet-dashboard-1.2.13-rc1.tar.gz Debian packages are available at https://apt.puppetlabs.com RPM packages are available at https://yum.puppetlabs.com See the Verifying Puppet Download section at:

Hi, I fresh installed Fedora Core 6. I downloaded and untar the 'asterisk-1.2.13' into /usr/src/asterisk-1.2.13. When I run ' make' I get: ... ... chan_phone.c:41:29: error: linux/compiler.h: No such file or directory make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.2.13/channels' make: *** [subdirs] Error 1 [root@sss asterisk-1.2.13]#

All, I have upgraded by home machine from Fedora Core 5 (FC5) to the recent FC6 and am struggling to build Zaptel-1.2.10 and Asterisk-1.2.13 on the box... which is an Intep P4 2.8GHz HT processor box with 845 chipset, hence the kernel installed is 2.6.18-1.2798.fc6-i686 so we hve this: [root@gate zaptel-1.2.10]# uname -a Linux gate.tubby.org 2.6.18-1.2798.fc6 #1 SMP Mon Oct 16 14:54:20

Hi all I have downloaded 1.2.13 installing on my FC5 when iam making, iam getting the following error could some one suggest me the what is the problem make[1]: Entering directory `/root/vici/asterisk-1.2.13/apps' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=k8

Puppet Dashboard 1.2.13 is a maintenance and bugfix release of Puppet Dashboard. This release is available for download at: https://downloads.puppetlabs.com/dashboard/puppet-dashboard-1.2.13.tar.gz Debian packages are available at https://apt.puppetlabs.com RPM packages are available at https://yum.puppetlabs.com See the Verifying Puppet Download section at:

Hi there I am using fedora21 for hosts and guests, using lxc containers, and was on vanilla 1.2.9 as exposed by the fedora yum repo The other day I was needing the ability to grant mknod capability to some guests, some chap on IRC kindly pointed me at the <capabilities> tag However I was told that needed 1.2.11 So I went and rebuilt from this download

Hi I've had an asterisk setup for the past 15 months, based on the debian asterisk packaging. Until late August of this year, I had no problems once initial setup was complete- the system worked essentially flawlessly. Since August I have been having exceedingly infuriating intermittent problems that are causing me occasional periods of nasty trouble: 1. No Dial Tone. Every Sunday night at

Dear people, is it possible to have presence using Asterisk 1.2.13 / SIP with Linux/Debian Etch??? I'd like to see if my intranet contacts are available, busy, disconnected.... Thanks a lot Alejandro

No matter what I do, ${ANSWEREDTIME} is always 0, even on the most simplest dial plan such as: Using Asterisk 1.2.13 exten => 77,1,Answer exten => 77,2,Playback(custom/dax/S300) ; one minute file exten => 77,3,Noop(${ANSWEREDTIME}) exten => 77,4,Hangup -- Executing Answer("SIP/5402-b7b45f58", "") in new stack -- Executing

The Asterisk Development team has released an update to Asterisk 1.2, Asterisk 1.2.13. This release contains a fix for a security vulnerability recently found in the chan_skinny channel driver (for Cisco SCCP phones). This vulnerability would enable an attacker to remotely execute code as the system user running Asterisk (frequently 'root'). The exploit does not require that the

The Asterisk Development team has released an update to Asterisk 1.2, Asterisk 1.2.13. This release contains a fix for a security vulnerability recently found in the chan_skinny channel driver (for Cisco SCCP phones). This vulnerability would enable an attacker to remotely execute code as the system user running Asterisk (frequently 'root'). The exploit does not require that the

The Asterisk development team is pleased to announce the release of Zaptel 1.2.13. This release contains a large number of bug fixes, an important performance improvement for most Digium cards, and support for new Digium hardware and some significant improvements in the XPP driver for Xorcom's Astribank hardware. In detail: * A modification was made to the drivers for all Digium PCI cards

The Asterisk development team is pleased to announce the release of Zaptel 1.2.13. This release contains a large number of bug fixes, an important performance improvement for most Digium cards, and support for new Digium hardware and some significant improvements in the XPP driver for Xorcom's Astribank hardware. In detail: * A modification was made to the drivers for all Digium PCI cards

Hi, After upgrading from: Zaptel 1.2.9.1 Asterisk 1.2.12.1 with bristuff-0.3.0-PRE-1s to Zaptel 1.2.10 Asterisk 1.2.13 with brustuff-0.3.0-PRE-1v I get the following error when connecting my Xlite Softphone: --- cut --- Nov 4 17:33:45 WARNING[4430]: chan_sip.c:1090 __sip_xmit: sip_xmit of 0x886df58 (len 486) to 192.168.9.9:31308 returned -1: Operation not permitted --- cut --- It seems to be

Hey folks, I recently upgraded my mail server from an older OpenBSD -CURRENT snapshot to the latest (Aug 1). As part of this upgrade, I updated all of my packages to their latest versions. Since this upgrade, dovecot has begun failing to authenticate intermittently. I use it primarily for web mail (Squirrelmail 1.4.20) and leave myself logged in during the day. This problem manifests

Hi, I think I need to ask you for advice. Personally I face that for the first time and hardly imagine this could be possible at all, but... My setup is simple: I have one mail server where all of my mail accounts are stored, and another one to store backup of all mailings. The scheme is simple, SMTP on first server forwards a copy of each message to address catchall at

Hi guys, I'm running Asterisk 1.2.13 on a Debian Linux system (that was just the version that was packaged for it). I've been using monitor() to record calls, with fairly satisfactory results - at least until the last few months. I've been recording VoIP calls, and using monitor() with no arguments, so I'm getting separate wav files for each leg (both use ALAW, BTW), and

Hello, I have the following problem. I am migrating my asterisk infrastructure to a new server and I encounter a strange problem. The configuration is as followin: IAX clients connect to asterisk which forward calls to a sip box connected to a phone line. On the old server everything works ok but on the new server, even if the logs are identical it seems like the dtmf number does not get passed

Hello I am trying to build libvirt 1.2.13 (latest) from source on a Ubuntu 14.04 64 bit box. After installing all the dependencies (libyajl, libdevmapper, libpciaccess, libnl), I could finish the build and install. However, invoking libvirtd throws this: root@ubuntu:/home/hvishwanath/Downloads/libvirt-1.2.13# libvirtd libvirtd: /usr/lib/libvirt-qemu.so.0: version `LIBVIRT_QEMU_1.2.3' not