similar to: Asterisk disconnected suddenly

Facter 2.0.0rc1 is a feature release candidate with bug fixes, features and other improvements. Facter 2.0 is designed to accompany the upcoming Puppet 3.0 release and has breaking changes from Facter 1.6.x. Facter 2.0 is not backwards compatible with Puppet 2.6 or 2.7. It includes contributions from the following people: Chris Price, Daniel Black, Daniel Pittman, Garrett Honeycutt, Gary Larizza,

The Asterisk Development Team has announced the release of Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases have been created in response to a SIP remote crash vulnerability. Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an SDP regression

The Asterisk Development Team has announced the release of Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases have been created in response to a SIP remote crash vulnerability. Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an SDP regression

Would someone please help with this error. Using: dovecot-1.0-2.rc28.fc6 On: Fedora Core 6 130 users all connect remotely via ADSL to server to pop their email. None of them have any problems at all with pop3. 3 users in one office, connect to the server via a NAT'd ADSL. Intermittently those three users will get a timeout error, Logfile shows: Apr 19 12:50:53 mail dovecot:

Hi everyone, I'm trying to make a cross-compiler based on llvm to target sparc (solaris) from x86_64 (linux). I couldn't compile llvm-gcc on my system so have to use the precompiled binaries from the website, but already have a gcc cross-compiler so can use this. So far, my compilation steps are these: 1) Preprocess the (C) sources using gcc cross-compiler 2) Use llvm-gcc to

On Thu, Apr 5, 2018 at 10:48 AM, Artem Russakovskii <archon810 at gmail.com> wrote: > Hi, > > I noticed when I run gluster volume heal data info, the follow message > shows up in the log, along with other stuff: > > [dht-selfheal.c:2328:dht_selfheal_directory] 0-data-dht: Directory >> selfheal failed: Unable to form layout for directory / > > > I'm

Hi! Since we upgraded to 1.6.1.11, asterisk is only outputing monitored files -in and -out. It is not mixing them in the end. queues.conf has monitor-type=MixMonitor... Would somebody help me debug why it doesn't mix the sounds?? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've got it ! Actually, it was obvious. A question of rights on the USB port. As soon as the process has the right rights (?) on the port, it runs smoothly. Sorry for the useless post... Regards, Pierre Boullier ----- Original Message ----- From: Pierre Boullier To: nut-upsuser at lists.alioth.debian.org Sent: Friday, November 12, 2010 3:18 PM Subject: usbhid-ups and APC Smart-UPS 1500

>> I have mounted a [homes] share on a Samba 3.0.22 server >> from my XP Pro PC. On my Samba server, I have set >> deadtime to 1 minute. After connecting, I execute a net >> use command and see that the status of the share is OK. >> After a minute or so with no activity, I see that the >> status changes to Disconnected. At this point, I thought >> that

I have mounted a [homes] share on a Samba 3.0.22 server from my XP Pro PC. On my Samba server, I have set deadtime to 1 minute. After connecting, I execute a net use command and see that the status of the share is OK. After a minute or so with no activity, I see that the status changes to Disconnected. At this point, I thought that when I re-accessed the share, I would be prompted again for a

Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the

We just started running Dovecot this summer at our campus. Overall things have gone very well, and performance have improved since moving off of U of Wash. We have run into an issue with a user running Thunderbird 2.0.0.6 on Solaris. He is seeing a very long response time (20mins+) on his first mailbox selection each morning when he comes into work. I looked at the Dovecot log, and the only

Hello All, I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The softphone can call the other one but I can' t hear any voice. My configuration files are below: [root at localhost asterisk]# cat sip.conf [general] canreinvite=yes [1001] username=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend

Hello! I'm trying to registers events of queues in /var/log/asterisk/queue_log and Mysql database .I have configured realtime queue_log on MySQL and works well, but /var/log/asterisk/queue_log file is empty, since you're not registering events of queues. Removing extconfig.conf configurations (queue_log => mysql,general), /var/log/asterisk/queue_log works well, events logs on

After upgrading to Asterisk 1.8.0, I am finding that my outbound SIP calls are being destroyed after 1 minute and 20 seconds (80 seconds). Asterisk is sending a BYE message - I have no idea why. http://jeremy.kister.net/tmp/20101115/sip.txt for a sip debug. can anyone suggest how i can further deal with this? -- Jeremy Kister http://jeremy.kister.net./

Hi, I'm trying to create a link between two PBXs. One is Asterisk 1.4.15, the other is an unknown 3rd party PBX. In my internal testing, beween two A*k servers, I found that if I created two sip accounts from the same IP, one as peer and one as user (intending to give an -IN and -OUT setup), then inbound calls always seemed to route via the -OUT account and failed. My fix was to use

Hi, I'm banging my head over this. Usually, I'm using a SIP hardphone feature called "Call Pickup Starcode" to enhance BLF with Directed Call Pickup : basically, SIP hardphone (here a Thomson ST2030S) is configured to send an INVITE message whenever a BLF is pressed while blinking. The INVITE is build with the extension number (attached to the BLF that was blinking and pressed)

Hi, I'm seeing weird behavior with AMI where no events are output until some input is detected (can be an empty line), at which time all the buffered output is spewed out at once. I am maintaining multiple Asterisk installations, and with one installation I have run into a weird buffering problem with AMI. The version is 1.6.1.11 in this particular case, which I am running at multiple

Ladies and Gentlemen, I have been playing around with Arnaud's new HAL stuff a bit, and have some questions and observations. First my configuration: my UPS (Peter may remember): ups.mfr: APC ups.model: Back-UPS ES 650 ups.productid: 0002 ups.firmware: 818.w1.D ups.firmware.aux: w1 kernel: 2.6.18-92.1.6.el5 hal-0.5.8.1-35.el5, nut based on nut-2.2.2.tar.gz, small changes to

I haven''t updated rails today and all of a sudden scaffold is pluralizing my names ruby script/generate scaffold Admin::Category produces class Admin::CategoriesController < ApplicationController Do I really want "CategoriesController" instead of "CategoryController"? Thanks, Peter