similar to: CALEA enforcement guidelines according to Comcast

On 16/12/14 18:15, Alexander Dalloz wrote: > Am 16.12.2014 um 16:03 schrieb For at ll: >> On 15.12.2014 12:50, Steve Clark wrote: >>> On 12/15/2014 05:51 AM, For at ll wrote: >>>> Hi >>>> >>>> I had a two repo for cento6 where I can download httpd 2.2.29, >>>> (baseurl=http://centos.alt.ru/repository/centos/6/$basearch/) and

I am interested in configuring Dovecot's TLS so as to retain forward secrecy, but eliminate all of NIST's elliptic curves. Besides being subject to side channel attacks [1], in some quarters there is a general distrust of NIST's curves and any of their other cryptographic primitives after the Dual EC DRBG debacle. >From what I can tell, the following will prevent the use of

Hi All, I was just reading through Info Week while on a flight and happened upon an brief piece about a new VOIP security intiative worked up by a handful of the usual suspects; Alcatel, SMU, NIST, Symantec, etc. All of this begs the question of can't we get just do this as a user community? I understand that the Zultys phone, which I own several, support AES encryption of the RTP stream.

Hi Timo, reading this http://www.kuketz-blog.de/perfect-forward-secrecy-mit-apple-mail/ it looks like DHE-RSA-AES256-SHA:DHE-RSA-AES128-SHA can be forced in use with apple mail ( if no ECDHE is possible ,by missing openssl 1.x etc, seems that apple mail tries ECDHE first if fails its going to use RSA-AES128-SHA ) force soltution as tried ssl_cipher_list =

Hello! Most are probably bored seeing another letter about this, but I've put in a fair amount work on a spec for rewriting the CDR system in Asterisk, and I have some questions: First, please look at what I've written so far: svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs and look at the file "CDRfix2.rfc.txt" in the RFCs dir. The spec SIGNIFICANTLY alters the way

Dear helpers, I'm trying to extract certain rows from a matrix according to the values the rows have in a certain column. I've been googling for a while without result. Here's a reproducible example of a matrix (and the one I was playing with initially): > myrepo<-getOption("repos") >

>From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop working, failing with chan_sip.c:4083 retrans_pkt: Hanging up call 7238b48c11581d4166b899bf747a05f7 at 130.211.62.184:0 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). is there any way to configure to have the previous behaviour? Im trying to set

On 12/2/2014 1:32 AM, Reindl Harald wrote: > > Am 02.12.2014 um 06:44 schrieb Will Yardley: >> On Mon, Dec 01, 2014 at 09:27:48PM -0800, Darren Pilgrim wrote: >>> On 12/1/2014 4:43 PM, Will Yardley wrote: >>>> Can you use both ssl_protocols *and* ssl_cipher_list in the same config >>>> (in a way that's sane)? >>> >>>> Is there a

Hi All First the essentials: dovecot --version: 2.2.15 /usr/local/etc/dovecot/conf.d/10-ssl.conf: ssl = required ssl_cert = </usr/local/openssl/certs/mail.domain.com.chained.dovecot.ecdsa.crt ssl_key = </usr/local/openssl/certs/mail.domain.com.ecdsa.key ssl_protocols = !SSLv2 !SSLv3 ssl_cipher_list =

gtgbr@gmx.net wrote: > Aaron Gaudio wrote: > >>Where is it released? Telling me I have to set up cvs to download > > > http://www.xiph.org/~msmith/ > > This has been said before. This is no good. This is by far not the first time someone has pointed out, that icecast2 is not mentioned _anywhere_ where one would expect to find it. By this I mean locations like

Hi guys I've been browsing the code and at many places I found the following odd sequence: char * string=malloc(somesize); ? bzero(string,strlen(string)); free(string); I really don't see why you would zero a string and free the memory immediately afterwards? Any idea why this is done? Thanks! Met vriendelijke groet Best regards Bien ? vous Miguel SANDERS ArcelorMittal Gent UNIX

I assume that from your previous post that you are using iconnect Is your register line in the format: Register => 18005551212:1234@213.137.73.178/18005551212 I've had good luck using the IP address vs. the fully qualified hostname. Remember that the register line goes in the [general] section of sip.conf. Also, are you using the latest CVS release of *? -----Original Message-----

Hi List; Where I can download Asterisk GUI and what I can have benifit from it? Regards Bilal ____________________________________________________________________________________ Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545469

Use oh323. Download the openh323 and pwlib tarballs from openh323.org Follow Jeremy's instructions in the /asterisk/channels/h323/ directory EXACTLY! good luck Regards, Sean Langley, P.Eng Firmware Engineer General Dynamics Canada (403)730-1482 sean.langley@gdcanada.com > -----Original Message----- > From: Senad Jordanovic [mailto:senad@cwcom.net] > Sent: Friday, September 12,

Hi, Companies that offer telephone service to the public are obliged to offer tapping to all kind of authorities. Does anyone know how to tap in Asterisk? I.e. record (or copy) a conversation based upon their telephone number? Thanks Dan _________________________________________________________________ Stay in touch with absent friends - get MSN Messenger http://www.msn.co.uk/messenger

Hi all, Anyone has experience using * through 128 kbs (or bigger) satelite link? In particular I am interested to hear how many calls could be put through 128Kbs satelite link simultaneously? Ta SJ

hi all we want to use Two single port Bri cards in Trixbox. Any idea which card is having good support and performance repotation especially when using two or more in Trixbox. Regards farooq --

Are there any nice GUIs out there for Asterisk Realtime? Google doesn't yield much. I spent a day trying to get VoiceOne to work without much success. Thanks, Mike Clark

Hi I have been playing with trying to get voicemail.conf from DB, I am using cvs-head, but when I start asterisk, it dies a horrible death, because it cant load any voicemil setting. I looked at my mysql logs to see what query was being sent, and I get SELECT category, var_name, var_val, cat_metric FROM voicemail_users WHERE filename='voicemail.conf' and commented=0 ORDER BY

Hi, I have enabled ztdummy in order to have * compile it. I can modprobe ztdummy with no problems. The sole reason, i need ztdummy is to heve musiconhold and meetme working. However when I start *, it says this and does not start. ---------------------------------------------------------------------------- ---------------------- == Parsing '/etc/asterisk/zapata.conf': Found