similar to: Logwatch for postfix

Hi, I''ve got a newbie question--sorry if this is covered elsewhere, I parsed through the archives for awhile and didn''t see it. I''d like to listen for whenever a file is renamed (e.g. foo.txt -> foo.old) and then magically change it back. This sounds odd, but I''m working with a stubborn application and this will actually make things work nice. So, if I do:

Hello there! During some research on Internet I found the following comparison on site Voip-Info (see, "http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ"): The main points listed on Asterisk's "CONS" that concerned me were: * Conferencing on Asterisk depends on Zaptel hardware and/or kernel modules for timing; * Lack of built-in STUN support for SIP NAT

Hello, I have error on compiling time I tried on HP-UX 11.00 system with gcc version 3.1 to compile samba source Samba latest version source was dowloaded from www.us1.samba.org/samba/ftp I attached log file with this error thank you. begin 600 samba_make.log.doc M(R!M86ME#0HO=7-R+VQO8V%L+W-A;6)A+V)I;B(@+410241$25(](B]U<W(O M;&]C86PO<V%M8F$O=F%R+VQO8VMS(B

Hello there! We are planning to use Asterisk on our VoIP platform, and we are spending some brains on a way to provide the following facility: let some SIP user (extension) registrate with more than one client (ATA, SoftPhone, VoipCelular, etc) - what isn't a problem at all -, initiate calls from any of this devices that are registrated with the same user - no problems on tests too -,

Hello there! I'm testing "Dial" call limit option on Asterisk version 1.4.26, but it's not working. The issued command is: "Dial(SIP/${EXTEN}|20|RtT|L(300000:60000:20000))". Am I missing something ? Does it only work with Asterisk version 1.6.X ? Thanks and best regards, -- __At.,

Hello there! I was testing Asterisk for the last two weeks using the Realtime driver for MySQL, and leaving "rtcachefriends=yes" configured to enable MWI. Today I started making additional tests with "rtcachefriends=no" because we will probably need to use Asterisk without this cache. For some strange reason, calls stop to get routed between the SIP clients. I've

Hello there! I need some help to configure two Asterix boxes to route calls using SIP. I followed the instructions present at this site: "http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html", but I couldn't get it working so far. The only difference, besides the names that I've used, is that I'm using realtime to retrieve

Hello there! I'm working on some modifications on Asterisk to adapt it to our needs considering some particular demandings of the infraestructure we want to provide. Two of these modifications are: 1- A proprietary configuration driver that will communicate with a server that will be the source of information for the entire infraestructure; and, 2- A call control application that will be

I made a few updates to OggPCM2 http://wiki.xiph.org/index.php/OggPCM2 reflecting the latest discussions. Could everyone have a look at it and see if they agree. Otherwise, what do you feel should be changed? Anyone wants to speak in support of chunked PCM? For all those that are just tired of this mess like me, please express yourself in the new spec I created: OggPCM3

Hi, I have posted this three times to the flac-dev, vorbis-dev, and ogg-dev mailing lists. I wanted to see what code there was currently to support Ambisonics. So I downloaded the code from the xiph download page for libogg-1.1.3, libvorbis-1.1.2, vorbis-tools-1.1.1 and flac-1.1.4, but wasn't able to find anything. If it exists then I missed it, so could somebody please point me to it.

On Tue, 31 May 2016 09:41:37 -0700 Michael Graczyk <mgraczyk at google.com> wrote: > UHJ is an interesting way to preserve compatibility with non-ambisonic > playback systems. However, I have not seen it generalized to higher > orders. I expect that its popularity will decrease as HOA becomes more > and more common. If UHJ becomes popular in the future, we could > specify

All, using Samba as an AD (2k12) domain member in Stretch (2:4.5.12+dfsg-2+deb9u4) with tdb as default and rid as domain backend. No overlapping. Everything works fine. Setup was done as in the wiki [1]. If you're connecting from a Windows 10 client and do not add dedicated keytab file = /etc/krb5.keytab kerberos method = secrets and keytab winbind refresh tickets = Yes to

The blowfish crypt(3) mechanism supports the use of a "cost value" for password encryption. The cost value is encoded into the encrypted password that is stored in master.passwd. On OpenBSD, this cost value can be set in login.conf. FreeBSD does not currently support the cost value. The cost value is the base-2 logarithm of the number of rounds of encryption to use so

> One obvious thing that seems to be lacking is the granulepos mapping. As > suggested in Ogg documentation, for audio a simple sampling frame number > ought to suffice, but I think the convention should still be spelled > out. I was under the (maybe wrong) impression that the Ogg spec already covered everything that's needed for granulepos. If that's not the case, please

Hello, I'm the maintainer of a syntax highlighting tool highlight.js[1]. Recently the Kaggle project has announced they wanted to sponsor the development of the R highlighting definition for it[2]. I wanted to drop a line about it here since I suspect this list has much more R programmers on it than our small discussion group :-). So if you're interested and don't mind touching a

On 2005-11-16, Jean-Marc Valin wrote: > Otherwise, what do you feel should be changed? One obvious thing that seems to be lacking is the granulepos mapping. As suggested in Ogg documentation, for audio a simple sampling frame number ought to suffice, but I think the convention should still be spelled out. Secondly, I'd like to see the channel map fleshed out in more detail. (Beware

If we are on the subject of byte compilation, let me bring a couple of examples which have been puzzling me for some time. I'd like to know a) if the compilation will likely to improve the performance for this type of computations, and b) at least roughly understand the reasons for the observed numbers, specifically why x[i]<- assignment is so much slower than x[i] extraction. The loops

Martin, I do not believe that there is any need for code specifically supporting Ambisonics. FLAC supports conversion to and from other audio formats which hold multiple channels. Ambisonics B-format involves channels WXYZ, C-format is UHJ or BHJ, G-Format can be any number of channels, but typically 5.1 on DVD. You can work with Ambisonic recordings by either using individual mono

On 3/22/07, Brian Willoughby <brianw@sounds.wa.com> wrote: ... > But to return to your question, exactly what kind of "support" are > you looking for? I wasn't looking for any particular support, but just to see what support was there (if any). ... > I do not believe that there is any need for code specifically > supporting Ambisonics. FLAC supports conversion to

Luke, Thank you for sharing the benchmark results. The improvement is very substantial, I am looking forward to the release of the byte compiler! The arithmetic shows that x[i]<- is still the bottleneck. I suspect that this is due to a very involved dispatching/search for the appropriate function on the C level. There might be significant gain if loops somehow cached the result of the initial