Displaying 20 results from an estimated 2000 matches similar to: "Update on sshfp 4"
2014 Apr 07
1
Ed25519 keys in SSHFP RRs
Hello.
Subramanian Moonesamy has gotten the ball rolling to include Ed25519 in
IANA's registry for SSHFP key types [1].
I've opened a bug report [2] that includes a patch that adds the needed
support code and provisionally assigns Ed25519 a value of 4 (values
1,2,3 reserved for RSA, DSA, and ECDA, respectively) [3].
The enhancement request/bug is meant to keep the issue on the radar.
2014 Apr 07
4
[Bug 2223] New: Ed25519 support in SSHFP DNS resource records
https://bugzilla.mindrot.org/show_bug.cgi?id=2223
Bug ID: 2223
Summary: Ed25519 support in SSHFP DNS resource records
Product: Portable OpenSSH
Version: -current
Hardware: All
OS: All
Status: NEW
Severity: enhancement
Priority: P5
Component: ssh
Assignee: unassigned-bugs at
2015 Apr 27
1
Development version of R: Improved nchar(), nzchar() but changed API
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
2014 Apr 21
1
Vorbis vs Opus
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
2012 Jul 17
1
[Bug 1972] ssh-keygen fails to generate SSHFP for ECDSA but exits with 0 code
https://bugzilla.mindrot.org/show_bug.cgi?id=1972
Daniel Black <daniel.black at ovee.com.au> changed:
What |Removed |Added
----------------------------------------------------------------------------
CC| |daniel.black at ovee.com.au
Keywords| |openbsd, patch
--- Comment #2
2018 Jun 12
2
T-38 re-invite issue
>>>>> "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
2016 Jan 27
4
PJSIP Stun/ICE
>>>>> "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.
2016 Jan 29
2
PJSIP Stun/ICE
>>>>> "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
2018 Jun 13
2
T-38 re-invite issue
>>>>> 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>
2014 Apr 25
1
srtp/dtls when sip is clear over lo
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:
2014 May 22
0
FollowMe reinvites
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
2014 Nov 23
0
Dahdi fxo vs sip blf
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
2015 Mar 03
0
TLS, SRTP, Asterisk11 and Snom870s
>>>>> "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
2015 Mar 03
0
TLS, SRTP, Asterisk11 and Snom870s
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
2003 Mar 05
3
IPv4...NAT...etc
http://lists.digium.com/pipermail/asterisk-users/2003-March/008088.html
"why is there such a delay in getting ipv6 rolled out when it solves all these problems ?"
===
There are many reasons...
1. Leasing Address Space from the I* society (small s...aka the Big Lie Society) is not desirable by all people...
2. The IPv6 Privacy Problem...that is especially important in the area of computer
2003 Aug 20
2
Strange happenings
Just idly watching * in console mode and saw that someone from
50.49.54.102 tried to register with my *.
whois gives:-
OrgName: Internet Assigned Numbers Authority
OrgID: IANA
Address: 4676 Admiralty Way, Suite 330
City: Marina del Rey
StateProv: CA
PostalCode: 90292-6695
Country: US
NetRange: 50.0.0.0 - 50.255.255.255
CIDR: 50.0.0.0/8
NetName: RESERVED-50
2005 Sep 06
4
Paranoid Firewalling
After reading this article:
http://www.theregister.co.uk/2005/08/31/blocking_chinese_ip_addresses/
I got to thinking that there is really no reason for *any* traffic to
hit my servers that comes from anywhere outside North America. So I
wrote the perl script at the end of this posting to extract selected IP
ranges posted at iana.org and convert them into iptables rules blocking
any traffic
2011 Jan 24
1
ECDSA and first connection; bug?
Folks,
I read the 5.7 release announcement and updated, to try out ECDSA. Most
parts worked very smoothly. The inability to create SSHFP records is
understandable, since IANA haven't allocated a code yet.
One apparent bug: I think StrictHostKeyChecking=ask is broken for ECDSA.
% ssh -o HostKeyAlgorithms=ecdsa-sha2-nistp256 localhost
2015 Jan 13
1
opus Digest, Vol 72, Issue 4
Martin Leese wrote:
> Subject: [opus] MIME Types and File Extensions
> To: opus at xiph.org
> Message-ID:
> <CAAzqGd_uzR646Nsdt=O2HDxLOYE2=K=5n9UOHLr3Y4BGzdVasw at mail.gmail.com>
> Content-Type: text/plain; charset=UTF-8
>
> Hi All,
>
> On the Xiph Wiki page at:
> https://wiki.xiph.org/MIME_Types_and_File_Extensions
...
> Could somebody more
2004 Apr 12
0
RE: Asterisk-Users digest, Vol 1 #3408 - 12 msgs
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