similar to: Play CD!

WorkShop Virtualiza??o com Proxmox VE ------------------------------ <https://www.facebook.com/sharer/sharer.php?s=100&p[url]=http://blog.konnectati.com.br/arquivos/1276> Dia 12 de Julho, S?bado, a partir das 08hs00 acontecer? o WorkShop Virtualiza??o com Proxmox VE. Venha conhecer uma das melhores ferramentas de Virtualiza??o do Mercado. Com este treinamento, sua equipe de TI ter?

Hi, I am using tin-0.3 and trying to route over a tinc tunnel. Here is my details: 192.168.10.0/24 / \ | | ethernet network | \ / ------------------------- | 192.168.10.254 (eth0) | | Linux Router | | 192.168.0.1 (tap0) | ------------------------- / \ | | | tinc tunnel | |

Hi, we're running Asterisk 1.2.10 and have been randomly being left blank voicemails with long messages that we can't hear. I've searched and searched but cannot find a solution. This is what happens: Internal Server runs Asterisk 1.2.10 where our mailboxes are Incoming Server (behind a firewall) runs Asterisk 1.2.13 and calls are bridged between this server and our internal server.

Hello, I have a table that was constructed in a wrong way (dput data on bottom - wrong data-frame): Local Mês Dia Colonia X6h X7h X8h X9h X10h X11h X12h X13h X14h X15h X16h X17h 1 Conceição Junho 1 3 2.16137 2.20412 2.08991 1.72428 1.69897 1.62325 1.44716 1.51851 1.43136 1.47712 1.51851 1.04139 2 Conceição Junho 2 3 2.46538 2.13672

This happens on configure, I only ran ./configure. This error is logged to stderr: [root at host1 dovecot]# wget https://dovecot.org/releases/2.2/dovecot-2.2.27.tar.gz [root at host1 dovecot]# tar xzf dovecot-2.2.27.tar.gz [root at host1 dovecot]# cd dovecot-2.2.27 [root at host1 dovecot-2.2.27]# ./configure --with-ldap=yes --with-zlib --with-bzlib --with-solr > foo configure: error:

Hello All, Sorry to rehash a question I am sure has shown several time but I cannot google up the answer from the lists. Does anyone know where I can get some royalty free, cost free music for my music on hold? I saw someone's post several weeks ago that said that this exists at a download site but I have not been able to find it. Thanks! Wiley Siler -------------- next

Hello All, I have Polycom IP 500 phones which I would like to have message waiting indicators on. So far, I have my system running well but the problem I am seeing is that MWI doesn't seem to tell my phone that it should display a MWI state. The light does not show when you have message nor is there any indicator on the text lines of a message waiting. The wiki doesn't cover this

Hi, Tim Not even one ocurrence of "wins support" on my smb.conf -----Mensagem original----- De: Tim Kelley [mailto:tpk@r00tserverz.net] Enviada em: s?bado, 18 de outubro de 2003 11:09 Para: Reinaldo Brand?o Gomes Cc: samba@samba.org Assunto: Re: [Samba] Nmbd in a infinite loop, doing nothing and sucking 99%CPU (really) On Saturday 18 October 2003 9:03 am, Reinaldo Brand?o Gomes

Hello Haki I fixed this problem following the instructions in /usr/src/zaptel-1.0.9/README.udev. Regards Cec?lia -----Mensagem original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]Em nome de Erdem HAKI Enviada em: sexta-feira, 15 de julho de 2005 05:11 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Assunto:

So we need to resample them first? Any live resamplers? -----Original Message----- From: icecast-bounces@xiph.org [mailto:icecast-bounces@xiph.org] On Behalf Of Geoff Shang Sent: S?bado, 12 de Febrero de 2005 09:51 a.m. To: icecast@xiph.org Subject: Re: [Icecast] Icecast Newbie Questions Anton Krall wrote: > 1. Can icecast limit the bps of an incoming live source? for example, > limit

Dear Regards, Confirmed here, the AD schema is at version 69. Other settings: Domain Level and Forest Functional Level could downgrade to Windows 2008. We need to replace this DC MS by Samba, but keeping the same domain. Does anyone have any suggestions or tips? De: Marc Muehlfeld <mmuehlfeld at samba.org> Para: Ricardo Pardim Claus <ricardo.claus at yahoo.com.br>; "samba

I have Asterisk server providing phone service for my company. The server is behind a PIX-515 FW and is assigned a private address 192.168.11.X/24. With that said what is best to provide remote SIP phones (home offices) securely. If the solution is to put up another Asterisk server with a public IP address I am opposed to that. I am looking for the a secure reliable solution to set up remote SIP

Dear Marc, Thanks for initial contact. The file server that need to migrate is a Windows 2012 R2. He's just a domain member. Being Win 2012, I can face any problems in the migration? Once the shares are ready, I plan to use Robocopy to copy the files and also the permissions of users and groups from Windows to Samba. De: Marc Muehlfeld <mmuehlfeld at samba.org> Para: Ricardo

Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? Or similar setup to a SIP provider using g729 or g723

Dear Marc,In the field we have only simple objects (GPO desktop, file server, users and groups).In total, there are approximately 100 desktops +120 users. Only one site. How to create a new domain in the Samba with the same SID of the current domain? So I could create a new DC with the same domain and SID, and import objects (users and desktops accounts). De: Marc Muehlfeld <mmuehlfeld

I don't see it, and can't really think of a good reason why it would happen. This happens in configure? And the first thing you ran was configure, right? No autogen.sh or autoconf/automake? -bash-4.1# cat /etc/redhat-release CentOS release 6.8 (Final) -bash-4.1# uname -a Linux test-centos6.dovecot.net 2.6.32-642.6.1.el6.x86_64 #1 SMP Wed Oct 5 00:36:12 UTC 2016 x86_64 x86_64 x86_64

Ok Mark Michelson. Thank you very much! You answer tells me that I was in the wrong path trying to access information from SIP 183 message. I need to find a way to let the callee pass information/data to the caller, even before accepting the call. That is, send data during the ringing time. And in my case, there will be more than one callee ringing at same time. As ASTERISK will not forward each

Cool, thanks for debugging this issue and letting us know. We have a few patches to fix this issue: - Introduce freeze in IR: https://reviews.llvm.org/D29011 - Lowering freeze: https://reviews.llvm.org/D29014 - Fix loop unswitch: https://reviews.llvm.org/D29015 Bonus patches to recover perf: - Be less conservative in loop unswitching: https://reviews.llvm.org/D29016 - Instcombine support

Hello Nhadie, I had a very similar situation. My solution, even tough might not look very wise, solved my problem the way I needed. I repeated the GotoIftime command in the next line in my extensions.conf . Like this: GotoIfTime(22:00-23:59|*|30|jul?test,s,1) GotoIfTime(00:00-02:00|*|31|jul?test,s,1) Rgs, Marco Cordeiro -----Mensagem original----- De: asterisk-users-bounces at

Dear Marc,Actually the file server is just a domain member. Therefore, I believe that is more peaceful to migrate. Here we have 2 servers Windows 2012 R2. A file server and a DC.Both servers will be migrated to Samba.We are documenting and preparing to migrate them.Once the migration is ok, I'll post the comments here.Thank you!Greetings! De: Marc Muehlfeld <mmuehlfeld at