similar to: SPA3000 and callerID

Hi, I want to apologise for my bad english (I'm french). So I need to Back-up a lot of datas from a place to an another by the way of rsync with ssh naturally. A good question from my boss was : What happened when the mysql database is used by a user during the Backup ? Is the system stop the Backup or is it transparency for rsync ? Sorry if my question is stupid but I haven't

Hi, I'm trying to configure a network bridge over a bonding without any success. Here is my configuration: ifcfg-eth0: DEVICE=eth0 HWADDR=00:1E:C9:E8:3F:E5 ONBOOT=yes USERCTL=no MASTER=bond0 SLAVE=yes ifcfg-eth1: DEVICE=eth1 HWADDR=00:1E:C9:E8:3F:E7 ONBOOT=yes USERCTL=no MASTER=bond0 SLAVE=yes ifcfg-bond0: DEVICE=bond0 ONBOOT=yes BRIDGE=br0 USERCTL=no ifcfg-bridge0: DEVICE=br0

Hi, I'm translating the document How rsync works to understand exactly how rsync works when it finds modified files. I'm translating "the sender" and I notice a strange fact. I would like to take an example where a byte is missing in a file. In facts : The generator has created a cheksum for the first file to the sender. Then, a checksum is created for the local file beginning

Hello, We are currently working on a migration Samba 3 to Samba 4 and I would like to know if one or more books on Samba 4 exist ? Thanks you in advance, [image: Geovariances] Mickael GOYAT *Administration Syst?mes et R?seaux* Geovariances, 49bis avenue Franklin Roosevelt - 77215 Avon Cedex - FRANCE +33.1.60.74.74.58 | mobile: +33.6.27.89.52.83 | fax: +33.1.6422.8728 | www.geovariances.com

Hi, I read that rsync can find files that we renamed. So, I would like to know how it can do that, if it's really possible ? I think there is a link with the file-list but I don't understand how it can detect that a file named 'test' is the same than the file renamed 'test_bis' for example. I didn't test it so I don't know if my question is correct. Micka?l

Hi, I have a little problem with Cwrsync. I don't know if one of you has already use it but I don't understand why rsync can't create directories. The next error is dislpayed : "rsync: recv_generator: mkdir "/*****" <in *****> failed: Permission denied <13>" I use a remote rsync server launched with rsync --daemon. I created a module pointing to the

Hi guys, I need to update my rsync-2.6.8 to avoid that error. I found the patch that Wayne creates in the archives of the mailing list but I don't know how to apply it. Can someone help me ? I tried to download the 2.6.8-4 rpm version of rsync but it seems the rpm doesn't work on my SUSE 10.0. When I do my rpm -iv ... it only writes a line with "rsync-2.6.8-4" ... useless. It

Hello, I was wondering if there is a way top specify rsync to replace only different block in case of in-place update. When I rsync a huge binary file that change often to a Copy-On-Write filesystem for backing it up (ZFS in my case, but I suppose that btrfs will act the same way) ; rsync replaces the whole file even with '--inplace' and so snapshots are inefficient. If you update only

Hi everyone, I'm trying the new atxfer functionality. All seems to work fine at the beginning, but there is no audio between the party at the end of the transfer. Plus, after that, even normal calls won't work well (they can't hangup!). I'm using the Asterisk CVS from 2005-01-10 with Asterisk-OH323. Here is my dialplan: [default] exten => h,1,NoOp(bye) exten =>

Hi! I'd like to be able to park/pickup a call with h323. Parking the call is easy, using ParkAndAnnounce. But ParkAndAnnounce does not return the parkinglot in a variable. So, I can't retrieve the call later. To keep it simple, here is a simple scenario of what I want: The call is established from phone 307. I enter DTMF code "#1" on phone 307. The call is then parked. I

Hello everybody, I work, I try, with TM but I have a problem with some special words in french. I think this is due to the manner to transform PDF to text, but I'm not perfectly sure. Let's see to the example : findFreqTerms(tdm1,30) [33] "<U+F0A3>" "<U+FB01>n" "<U+FB01>nancement" "<U+FB01>nancier"

Hi, all I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have enabled gateway function in SPA3000 and

Hi, all A quick question. I have SPA3000 and trying to get call waiting to work. I do receive call waiting tone, however hook flash does not seem to work. I think, I set up SPA3000 correctly. Basically, doing HF 2 (switching calls in Australia) does not do anything. Is there any examples on how to setup hook flash operation in Asterisk? Thanks, Rudolf

Hi, These days I run into something quite odd. I have an A@H that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

Dear all, Can someone from Netherlands who has SPA3000 send me the Regional & Line1 settings. I'm not sure what is wrong but I can call from asterisk to phone attached to spa300 FXO, but not the other way. I tested three phones: siemens gigaset, tiptel 160 and hpoj k80(fax). Only tiptel 160 can call to the asterisk, other two just dial but nothing happens. I guess this has something with

Hi all, Can anybody explain what these values exactly mean. As you all know its the dialtone value on an SPA3000 of linksys. 350@-19,440@-19;10(*/0/1+2). Can anybody help me how to write this code for a dialtone of frequency 425 which is continous. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have a little situation here :( I have an Asterisk working, and in another office, a Sipura SPA-3000. I configured the SPA and I have the extension working, the incomming trunk working, but the outgoing trunk (peer) does not work. The issue is that I have a dynamic IP where the SPA is, and neither the SPA nor my router have DynamicDNS. So, if I

Hi, I'm using a SPA 3000 as FXO and FXS termination connected to my * box. I'm using the caller ID prefix trick explained here: http://www.voip-info.org/wiki-Sipura+3000. All seems to work really fine, there is only a problem when I hangup my IP phone after a conversation and the other party don't: in this case the PSTN line remains connected until the other party hangs up. Any

My porblem is incoing PSTN calls are being forwarde to the * box, the phone rings, but when the phone is picked up, the call is not taken - it continues to ring. I am forwarding the call to (<S0:6666>) in my dial plan Can anyone assist? This is driving my crazy! Extract from the * console Executing Dial("SIP/3001-047c", "SIP/2004") in new stack -- Called 2004 --