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...mkdir src ln -s /tmp/rsync_bug/mnt src/dir rsync -aHx --delete --numeric-ids src/ dst1 rm src/dir mkdir src/dir cp -a mnt/test src/dir/ rsync -aHx --delete --numeric-ids --link-dest /tmp/rsync_bug/dst1 src/ dst2 || echo RSYNC FAIL Output for me: buczek at theinternet:~$ sudo ./rsync-bug.sh + '[' -d /tmp/rsync_bug ']' + umount /tmp/rsync_bug/mnt + rm -rf /tmp/rsync_bug + mkdir /tmp/rsync_bug + cd /tmp/rsync_bug + mkdir mnt + mount -t tmpfs tmpfs mnt + ln -s BLABLA mnt/test + mkdir src + ln -s /tmp...

Hi, A, B and C are three machines. A and C directly access to theInternet while B access to the Internet through A. +-------------------------------------------------------------+ | +-------------+ +-------------+ | | | A | | B | | | | | --- eth0 ---> <--- eth0 --- |...

Good morning, I am trying to use the Windows iax client. My iax.conf looks like this: [general] port=5036 bindaddr=10.1.3.111 bandwidth=high allow=gsm ; Always allow GSM, it's cool :) tos=lowdelay [pos| type=friend context=default auth=plaintext secret=pos deny=0.0.0.0/0.0.0.0 permit=10.1.3.0/255.255.255.0 host=dynamic defaultip=10.1.3.2 In the registrations dialog

Hello, today I tried a DIAX -> * -> DIAX connection over the internet (768/128 ADSL connection on both sides). The sound quality was great. However, we had some latency problems, and also, if both sides where not talking the first words had some problems getting thru. Is this expected, is there anything that can be done on our setup, any magical iax.conf entry? Thanks and best regards

I have my asterisk box on the public network. I have a winders box on the public network, running diax. I have a winders box, same setup, behind my Linux iptables firewall, on a private network. Both boxes cann register iax2 to asterisk, and dial, but as soon as asterisk tries to do the native bridge, the call fails. I've poked at this way too much today... what ports do I need to open for

Hi, the app HasNewVoiceMail can't find my voicemail. This is the errormessage: Dec 3 14:24:01 NOTICE[12222481]: app_hasnewvoicemail.c:104 hasvoicemail_exec: Voice mailbox 25 at /var/spool/asterisk/voicemail/default/25/(null) does not exist however this is the output of lspbx:~# ls -l /var/spool/asterisk/voicemail/default/25/ total 316 -rwx------ 1 root root 11814 2003-11-22 18:18

Hi all, Does anyone know, if it's possible to reroute all (i.e. HTTP / FTP / DNS / SMTP / POP3 / IMAP / etc) from an ADSL connected machine via another server, which is currently hosted with IS and has full internet access? i.e. Can I setup another machine, on a different public IP than the dynamic ADSL IP as default gw? OR do I need todo something on that machine to work as a router for

Hi all. This might seem really naive, but can mtree be used effectively as a native-to-core-OS tripwire equivalent? Would it be as efficient in terms of time-to-run and resource requirements? What sort of pitfalls should I be aware of? Has anyone here done this? If so, would you care to share your scripts/techniques? Thanks, Dave -- ______________________

I'm constructing a Web server which may require restricted areas of the site to be used from public places where a password might be sniffed. The damage that could be done by taking snapshots of the content from one session with a spy program is minimal. What the owner of the server does NOT want, though, is to allow unauthorized parties to gain unfettered access by stealing the password via

Hi all, I have developed a full featured Windows IAX phone based on LIBIAX library . It is now in a prerelease version (0.9.0) and you can download it for free from my web page: http://www.laser.com/dante or http://www.geocities.com/tdanro Some of the features are: - registering with Asterisk PBX; - can use any audio device as ring device (including PC speaker), independent of the play device;

Hi all, As promise, the new prerelease (0.9.2) is now available for download from the followiing locations: http://www.laser.com/dante or http://www.geocities.com/tdanro A detailed help file is available online and in the application package as a chm file, accessible from the app help menu too. Unfortunately the IAX2 support is not ready yet, but I work on it now (next on my list). The DLL used

OK, an official OpenSSH advisory was released, see here: <URL: http://www.mindrot.org/pipermail/openssh-unix-announce/2003-September/000063.html > The fix is currently in FreeBSD -CURRENT and -STABLE. It will be applied to the security branches as well today. Attached are patches: buffer46.patch -- For FreeBSD 4.6-RELEASE and later buffer45.patch -- For FreeBSD 4.5-RELEASE and

Hi, after getting the chan_capi error messages sorted out, I have another question. Calling * via SIP produces very good sound. Calling * via the chan_capi produces horrible sound. However, if I dial 500 in the demo menu to connect to the IAX at digium the sound is good again. ie: ISDNCall->AVM-B1-Card->Asterisk = All prompts sound horrible SIP->Asterisk = Prompts are good

Is there a Debian package available for asterisk-oh323, or the chan_h323? If yes, where might I find one? Thanks rgds pos

Hi, I have asterisk behind my primary PBX connected via ISDN (chan_capi). Calling out and calling in works just fine, however I can't connect to my primary pbxs' extensions. If anyone has an example extensions.conf, I'd be grateful for a copy. I tried (the MSN of the ISDN card is set to 30) exten => 22,1,Dial,CAPI/30:22 but this does not work. Changing it to exten =>

Hi, in a HTML page I can write href="mailto:joe@doe.com" and clicking on the link will open the default mail application. Is anything like that possible with any of the soft phones (SIP or IAX [Windows])? Any and all information is greatly appreciated. rgds pos

Hi, I am contemplating moving asterisk onto my firewall/DSL router machine. Any thoughts security wise on this? Usually I don't like to put any applications onto the perimeter of my network, but * seems to be a good candidate because of NAT issues otherwise. Any and all thoughts are greatly appreciated. rgds pos

Hi, we have implemented a first version of call support from a web based system for Asterisk (via the manager interface) and other, callto: and tel:, providers. Now I am looking at the other way around. If a call comes in, I want our web based system to automatically detect the number and present the call information to the user. Ideas anyone? I guess, I won't be able to get this done

Good day, anyone out there running an ISDN pbx behind an Asterisk server with an AVM active card? At this moment Asterisk is connected to the internal ISDN bus of the PBX, but I want to control the calls coming in from the outside, before sending them of to the PBX. Any BTDT is greatly appreciated. rgds pos

Hi, is it just me, or are the VoIP providers for Germany more expensive than going via call-by-call? sipgate.de lists a price of 1.76ct a minute, a couple of call-by-calls are listed at 1.3ct-1.5ct a minute. Did anyone else thought this to be strange? What about other countries? Same thing?! It seems to me, VoIP providers only make sense for headquarter to subsidiary, but not for regular