similar to: Standalone server providing authenticated services to AD clients?

Hi, Earlier I was having libvirt-0.7.5 but now I have installed 0.9.8 version of libvirt now. I am trying to start libvirt daemon using the command below but daemon it is not getting started and the PID file is not getting created. Tried /usr/loval/sbin/libvirtd -d option too. In that case I am not getting any error but PID file is not getting created in this case too.

Just wondering if the decisions in the bnetd case might affect Samba at all? http://www.freedom-to-tinker.com/doc/2004/bnetd_30sep.pdf From my reading, the judge is claiming that because the bnetd authors didn't have permission to reverse engineer the Battle.Net protocol, they breached copyright by reverse engineering it. Apparently in the USA you can waive your legal rights through

Hi All ; One of our customer is suffered from FS being corrupted after an unattanded shutdonw due to power problem. They want to switch to ZFS. >From what I read on, ZFS will most probably not be corrupted from the same event. But I am not sure how will Oracle be affected from a sudden power outage when placed over ZFS ? Any comments ? PS: I am aware of UPS''s and

On Tue, 16 Jun 2015 11:35:26 -0400 sean darcy <seandarcy2 at gmail.com> wrote: > My asterisk server is in the cloud. Figuring out how to send an email > is too much brain damage. So i can't use the email feature that's > built into voicemail. Really? That was one of the first things I did when I learned Asterisk. It was dead simple. Rather than creating some sort of Rube

This might be a rube question; if so I apologize in advance. Someone asked me to help and I don't have much info, but I have a limited window and have to be forearmed with as much information as possible before I set foot in the door. The guy that asked for help runs a Red Hat 6.2 box with the version of Samba that came with RH 6.2. I asked him what version of Samba it was. He

Recently, we were notified that the mechanism included in our Asterisk source code releases to download and build support for the iLBC codec had stopped working correctly; a little investigation revealed that this occurred because of some changes on the ilbcfreeware.org website. These changes occurred as result of Google's acquisition of GIPS, who produced (and provided licenses for) the iLBC

Recently, we were notified that the mechanism included in our Asterisk source code releases to download and build support for the iLBC codec had stopped working correctly; a little investigation revealed that this occurred because of some changes on the ilbcfreeware.org website. These changes occurred as result of Google's acquisition of GIPS, who produced (and provided licenses for) the iLBC

On 06/16/2015 11:52 AM, D'Arcy J.M. Cain wrote: > On Tue, 16 Jun 2015 11:35:26 -0400 > sean darcy <seandarcy2 at gmail.com> wrote: >> My asterisk server is in the cloud. Figuring out how to send an email >> is too much brain damage. So i can't use the email feature that's >> built into voicemail. > > Really? That was one of the first things I did

Hi all, I wrote a small script (developed and testet on Solaris 8), which automates the generation and installation of the steps needed to put keys in place. I you are interested to take it, feel free to do it. -- *** Freundliche Gruesse **** Best regards *** Anton Burkhalter Dipl. El. Ing. HTL Mobile:+41(0)78 844-0290 mailto:anton.burkhalter at gmx.net

Le 17/04/2018 ? 14:18, kfx a ?crit?: > dovecot 2.2.34 > solr 7.2 > > I only see new messages after typing on the server "doveadm fts rescan > -u username" though I've followed the wiki and added "fts_autoindex = > yes" in 90-plugin.conf . Subsequent search for the same pattern always > gives the same result, ignoring new emails with that particular

Hello, Asterisk seems to try to authenticate incoming INVITE based on the [section] in sip.conf and not the username specified. I just removed the "insecure" option from my sip.conf requesting every connection to be authenticated. I added the match_auth_username=yes in the [general] section for extra security. To make it work, I have to use the same [section] identifier as username.

Hello, I'm trying to receive a fax to my inbound number from voipbuster. Asterisk receives the call and starts the rxfax application successful, but then nothing happens. The calling party is still hearing a ringing tone, or sometimes nothing. Voicecalls are working correct and without problems. For testing I've add a local number (300) to the dialplan. When I call this number

Il 20/04/2017 17:32, kevin.larsen at pioneerballoon.com ha scritto: > > This gets kinda Rube Golberg-ish, but convert the incoming analog line > to sip, route it through asterisk and have asterisk do its thing > before converting it back to analog to send to the phone. Only problem > is you get a lot of extra hardware involved in the mix to make it > work. It will be a lot of

I am clearly doing something ridiculously wrong. Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are unable to register. They keep trying and then time out. With the sip debug on in Asterisk nothing is logged. Here is the trace from one of the phones (kphone): (192.168.100.13 is kphone, 192.168.100.3 is Asterisk) sipclient: sending: 21:47:45.454

Hello Momchil, (CC’ing more people that could correct me if I’m wrong) Thanks for looking into this. More answers below: > On 9 Apr 2018, at 17:57, Momchil Velikov via llvm-dev <llvm-dev at lists.llvm.org> wrote: > > Hello, > > So, I have this testcase: > > void f(int n, int x[]) { > if (n < 0) > return; > > int a[n]; >

Hello, So, I have this testcase: void f(int n, int x[]) { if (n < 0) return; int a[n]; for (int i = 0; i < n; i++) a[i] = x[n - i - 1]; for (int i = 0; i < n; i++) x[i] = a[i] + 1; } that, compiled with -O1/-Os for AArch64 and X86, generates machine code, which fails to properly restore the stack pointer upon function return.

I created an account at IAXTEL.com to route 1-700-XXX-XXXX calls through. IAXTEL.com gave me a number (example) of 700-555-6226. I have made the following changes to my: /etc/asterisk/extensions.conf: [iaxtel700] exten => _81700XXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1}) exten => _81800NXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1})

How can I get a FXMDIChild to shrinkwrap to its content? Here''s what I''m doing now: def create(*) self.height = maxChildHeight + 30 self.width = maxChildWidth + 10 super end This works, but I''m not crazy about manually adjusting for (apparently) frame and title bar sizes. -- vjoel : Joel VanderWerf : path berkeley edu : 510 665 3407

Hi Guys, I have Spikko setup as provider of DID and outbound routes and I can make calls out but no inbound calls via DID can be made. I did a sip debug which is reported below. I never receive the call though, I have a catch all in my inbound routes and it doesn't hit my context at all or not sip invite comes in: FreePBX: Trunk Name: *Spikko* Peer Detail *username=MyUsername*

Hi list, I try to connect to a GW which have one domain eg sip.mydomain.com and have few IPs related to this domain. I register * to this domain with host=sip.mydomain.com and type=user. So DNS will decide on which IP of my domain I will register (or redirection on the GW side). If an incoming call arrive, I would guess that, as type=user, it will not try to match the IP from INVITE as I