similar to: dahdi channels busy/congested

Small addition again: This error shows up in the log while removing snapshots WITHOUT rendering the Vms unresponsive — Jun 01 01:33:45 mc-dc3ham-compute-02-live.mc.mcon.net libvirtd[1657]: Timed out during operation: cannot acquire state change lock Jun 01 01:33:45 mc-dc3ham-compute-02-live.mc.mcon.net vdsm[6839]: vdsm vm.Vm ERROR vmId=`56848f4a-cd73-4eda-bf79-7eb80ae569a9`::Error getting block

Hello Soeren. I've started to look at this issue and I'd agree that at first glance it looks like a libvirt issue. The 'cannot acquire state change lock' messages suggest a locking bug or severe contention at least. To help me better understand the problem I have a few questions about your setup. >From your earlier report it appears that you have 15 VMs running on the

Hi, I would send those, but unfortunately we did not think about the journals getting deleted after a reboot. I just made the journals persistent on the servers, we are trying to trigger the error again last time we only got half way through the VM’s when removing the snapshots so we have a good chance that it comes up again. Also the libvirt logs to the journal not to libvirtd.log, i would

Hi Adam, Hi Eric, We had this issue again a few minutes ago. One machine went down exactly the same way as described, the machine had only one snapshot and it was the only snapshot that was removed, before that in the same scriptrun we deleted the snapshots of 15 other Vms, some without, some with 1 and some with several snapshots. Can i provide anything from the logs that helps ? Regards

Hello everybody, I've been using asterisk 1.2 for quite a long time now, but I thought it's time to try a newer version of asterisk. So I downloaded 1.8.5, extracted the tar, ran configure, make, make install ... Everything looks fine (no obvious compile/link errors). But as soon as I start asterisk, it dies with a segfault. I executed asterisk within strace and last action before the

Yes, I had a typo in the mail, but not in the cronjob Still wondering how to get certbot-1.7.0-1.el7.noarch working on CentOS 7 again.

Not directly an answer to your question, but we had so many problems with the certbot in different constellations, that we moved to https://github.com/acmesh-official/acme.sh which works just fine basically everywhere cheers Soeren ?On 05.10.20, 15:18, "CentOS on behalf of Alexander Farber" <centos-bounces at centos.org on behalf of alexander.farber at gmail.com> wrote:

dahdi_cfg -vvv error `cut: '/sys/bus/dahdi_devices/devices/pci:0000:01:00.0/spantype': No such file or directory` PRI show spans --command not found test call not working. -- dahdi_genconf -vvv Default parameters from /etc/dahdi/genconf_parameters Generating /etc/dahdi/assigned-spans.conf cut: '/sys/bus/dahdi_devices/devices/pci:0000:01:00.0/spantype': No such file or

On 03/06/15 07:36 +0000, Soeren Malchow wrote: >Dear Adam > >First we were using a python script that was working on 4 threads and >therefore removing 4 snapshots at the time throughout the cluster, that >still caused problems. > >Now i took the snapshot removing out of the threaded part an i am just >looping through each snapshot on each VM one after another, even with

Hi All, I am a new bee to *. I just installed Asterisk@home on FC3. I hv a FXO card. I hv configured two extensions one x-lite and other iaxComm. I configured * using AMP. The following setup works - x-lite (x 200) to iaxComm (x 201) - PSTN to x-lite - PSTN to iaxComm Voice mail, weather etc work fine. When i try to make an external call i am getting message "All routes are busy". In

Dear all, we have a very strange problem, - we have 3 VPN endpoints - all are in one NETWORK - all daemons come up and connect without any problem and normally we have no problem working through the VPN but in some cases the connection does not work because the traffic leaves the TAP interface on one VPN endpoint but never arrives on the other end, the similarities between the packages seem to

Dear, I have this scenario: - PBX Asterisk 1.6.2.10 with private IP 192.168.0.10 - Behind a Cisco ASA firewall that connects to Internet - SIP trunk to Net2Phone with these parameters (nat=no): host=200.58.113.60 username=DOLLY secret=123456 port=5060 type=peer dtmfmode=rfc2833 disallow=all allow=alaw&ulaw nat=no canreinvite=no qualify=yes -Softphones Xlite The PBX can't register to

Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before

AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI (single span). I'm sure I just have something goofed up in the dialplans? I have a bunch of Polycom 331 IP phones connecting to the server. I can dial the other extensions in the system fine and I can dial long distance outgoing but cannot seem to get it to dial local (7 digit) calls. I see this in the CLI: --

Hi All, I have searched the various post and not able to find the solution. I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same. When i try from softphone i got error as "Call rejected" and in the asterisk i got error as

Does anyone know what this error message means? Unable to create channel of type 'DAHDI' (cause 0 - Unknown) I've upgraded to 1.6.0 with dahdi 2.0. For some reason my outbound dahdi calls are not going through. At some point, it starts to work, but I don't know what the trigger is. Out of the blue, outbound calls start to work. I had been using asterisk-1.6-beta9 with zaptel

What is involved in provisioning Asterisk to use a multiline analog service from our local telco? I will only have one twisted pair entering in on a OpenVox card but am not sure how Asterisk interprets and deals with two incoming calls and/or two outgoing calls? Thanks! jlc

i have a prablom here, i want to send a call from A to B use sip trunk , the call can sended B,but not work to PSTN. the message from B server. help pls,what's rong? > > <--- SIP read from 192.168.0.176:5060 ---> > INVITE sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport

Hi all, i am stuck with the configuration of asterisk - modules are loaded ( zaptel and wct4xxp ) - i have zaptel.conf configure, output of ztcfg -vv --- snip -- rapid:~# ztcfg -vv Zaptel Configuration ====================== SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet

Dear All, I have an asterisk server version 1.2.12.1 along with trixbox and I am having this nasty problem, I have a TE200P and have an E1 pri attached to it and zttool says it's OK, I have configured the whole 31 channels into one group as follow: Zapata-auto.conf: callerid=asreceived signalling=pri_cpe switchtype=euroisdn context=from-zaptel group=0 channel=>1-15,17-31