similar to: Another audio file

Does vorbis have any niches of technical superiority over opus? Or is compatibility with older hard- and software the only benfit? Put another way, is there any reason to prefer vorbis over opus for music on new sortware or platforms? -JimC -- James Cloos <cloos at jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

>>>>> "DC" == D'Arcy Cain <darcy at VybeNetworks.com> writes: DC> Perhaps someone can explain what t38timeout is supposed to do A 'git grep t38timeout' on the src leads one to res/res_fax.c, where one case see that it is the number of miliseconds to permit for t38 negotiation to complete once it starts. Ie after both sides select t38, until they

>>>>> "JC" == Joshua Colp <jcolp at digium.com> writes: JC> I disagree that it makes it worthless for a large number of JC> users. It's only within the last few days that a few people have run JC> into this particular issue where they have a public IP address that is JC> changing a lot and PJSIP does not support changing it without a JC> restart.

Dear Martin, Does the work on nchar mean that bugs #16090 and #16091 will be resolved [1,2]? Thanks, Mark [1] https://bugs.r-project.org/bugzilla3/show_bug.cgi?id=16090 [2] https://bugs.r-project.org/bugzilla3/show_bug.cgi?id=16091 On Sat, Apr 25, 2015 at 11:06 PM, James Cloos <cloos at jhcloos.com> wrote: > >>>>> "GC" == G?bor Cs?rdi <csardi.gabor at

The progress metre in scp(1) breaks when the tty is too wide. This patch is the effortless fix: ########################################################################### :; diff -u openssh-2.1.1p2/scp.c openssh-2.1.1p2+jhc/scp.c --- openssh-2.1.1p2+jhc/scp.c Thu Jun 22 07:32:32 2000 +++ openssh-2.1.1p2/scp.c Sat Jul 1 22:15:36 2000 @@ -1176,8 +1176,9 @@ i = barlength *

Hello: Thanks for the input so far. Heres the issue-- This is a production environment-- where many people "touch" the files. ie-- The audio engineer is a freelancer who wants to master the files at the highest quality TO HIS EAR and experience-- He knows NADA, Not a thing about SOX-- but is a ProTools GURU. The SOX resampled files work on our asterisk box-- but I gotta put someone

>>>>> "AS" == A J Stiles <asterisk_list at earthshod.co.uk> writes: AS> If you are paying for a business-grade Internet connection, you AS> should get a static IP address -- or a block of them -- as AS> standard. Maybe you need to change your ISP? In some places (including here) static ip is not affordable. -JimC -- James Cloos <cloos at

I just went to grab p2 to upgrade and couldn't find it from <http://www.openssh.com>. The OpenBSD/OpenSSH directory is empty on all of the OpenBSD mirrors I looked at, including ftp.openbsd.org. I had to follow the links from mindrot.org to find the old pages. <http://violet.ibs.com.au/openssh/files/> does point to usable locations for the portable version. -JimC -- James H.

Are there any known problems converting dtmf from oob over iax2 to inband over rtp/ulaw? Obviously it works when converting to inband over pri/ulaw et al, but how about rtp? I've got packet traces that confirm that 2833 packets are properly generated when I have 2833 configured for the rtp link, but the other side seems to be ignoring those packets. So I tried inband on that link; nothing

If you are using ethereal to decode packet traces that include iax2 packets, you may have noticed that codecs such as ilbc were being shown as unkown. I've had a patch accepted into the ethereal cvs that corrects that, updating packet-iax2.[ch] to match asterisk cvs HEAD. I presume it will be in the next release, and is now available in ethereal's anon cvs tree. -JimC -- James H.

When testing chacha20-poly1305, I noticed that aes-gcm is significantly faster than aes-ctr or aes-cbs with umac. Even on systems w/o aes-ni or other recent instruction set additions. And there seems to be consensus in the crypto community that AEAD ciphers are the way forward. As such, it promoting the AEAD ciphers to the head of the preference list looks like a good idea. That would mean

Before coding this in C, I wanted to test the idea out in R. But I'm unsure if the theory is well-founded. I have a (user-supplied) black-box function which takes R^n -> R^3 and a defined domain for each of the input reals. I want to send some samples through the box to determine an approximation of the convex hull of the function's range. (I'll use the library from

I am looking to install a web interface for Asterisk to transfer calls and look who's on the phone. If anybody has a working web interface please let me know. I installed the www.asternic.com (operator) But when I bring up my web browser it says transferring data and does not bring a browser. -----Original Message----- From: asterisk-users-admin@lists.digium.com

>>>>> D'Arcy Cain <darcy at VybeNetworks.com> writes: >> Ie after both sides select t38, until they agree on the t38 terms. > OK, so does that mean that setting it to 25000 should leave time for the > re-invite or does the timeout start after that. As I wrote above, after that. After the sip/sdp. -JimC -- James Cloos <cloos at jhcloos.com>

The IANA has pre-allocated id 4 for ed25519. If waiting on the IANA were a reason to delay applying the SSHFP_KEY_ED25519 patch, it needn't be any longer. I've proposed un-reserving hash type 0 to be a "NULL hash", for those who'd rather publish the public key unhashed. Even if zero for unhashed fails to gain traction, I hope to see something allocated for that. But

Given a box with a sip proxy listen(2)ing on 0.0.0.0 and chan_sip or chan_pjsip listen(2)ing on 127.0.0.1, with ast sending rtp directly, will ast negotiate srtp or dtls even ast and the proxy speak sip in the clear over the lo interface? Avoiding encryption over lo can aid debugging, but will doing so also block secure media? -JimC -- James Cloos <cloos at jhcloos.com> OpenPGP:

For a sip-only application, what exactly is required to ensure that calls completed via followme are reinvited? Can it at all? The code after outbound = findmeexec(targs, chan) calls ast_bridge_ call(). I don't see anything there which can cause a reinvite, yes? When the same peer is used for both the incoming and outgoing legs, it is a bit of a waste to proxy the rtp. And even when the

It has been may years since I've done anything with a dahdi fxo; much has changed in the interim and I havne't found answers googling. The fxo hw is installed on the pots line in parallel to existing pots phones. My goal is to have a blf on the sip phone which lights whenever any of the devices on the pots line are off hook and which, when pressed, INVITEs the asterisk box such that it

>>>>> "JBB" == James B Byrne <byrnejb at harte-lyne.ca> writes: JBB> tcpenable=yes JBB> tlsenable=yes JBB> tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt JBB> tlscafile=/etc/pki/tls/certs/ca-bundle.crt JBB> tlsdontverifyserver=yes JBB> tlscipher=ALL JBB> tlsclientmethod=tlsv1 You are missing the tls key. The config name is

Other things to consider: The transport config, which can be in [general] or in a peer's [] block. if you want tls-only, use transport=tls it also accepts tcp, udp or a comma-separated list. if given a list, it tries them in order If you need ast to register over tls, use something like this: register => tls://username:xxxxxx at sip-tls-proxy.example.org (copied from the