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On Friday, the IC3 (FBI/NW3C/BJA) put out a security advisory on their website that contained a fairly vaguely worded warning about Asterisk systems being compromised and then being used as "vishing" (voice phishing) platforms. They were non-specific on the threat other than to advocate upgrading t...

...0.087, 0.138, 0.132, 0.244, 0.354, 0.421, 0.606, 0.74, 0.818, 1.048, 1.229, -1.17, -1.159, -1.16, -1.143, -1.137, -1.085, -1.028, -0.965, -0.919, -0.942, -0.902, -0.828, -0.79) ################this is the function that is called by mle2 exfunc <- function(c1,c2,c3,cs1,cs2,cs3,ic1,ic2,ic3,cc1,cc2,cc3,alstar,beta1){ x<-matrix(0,years*cross,3) x[,1] <- 1 x[,2] <- ra x[,3] <- sa coeffs <- vector(length=3) coeffs[1] <- c1 coeffs[2] <- c2 coeffs[3] <- c3 csp <- rep(c(cs1,cs2,cs3),each=years) e1 <- (x %*% coeffs);frfact <- csp*plogis(e1)...

...ected to this span get NO DIALTONE for every ATD. Modems on other spans continue to operate. This error seems to appear mostly on span 5 and 6. After restarting asterisk everything is okay again for a while. Any hints about what is going on here are greatly appreciated ;-) TIA, Bruno - bruno @ ic3s.de ------------ Connected to Asterisk 1.0.7-BRIstuffed-0.2.0-RC8g currently running on pbx (pid = 20305) Verbosity is at least 5 pbx*CLI> show channels Channel (Context Extension Pri ) State Appl. Data Zap/25-1 (pri1 s 1 ) Up Bridged Cal...

...the password, like post-SP3 NT clients do? (I used the EnablePlainTextPassword option on my NT boxes to take care of that, btw) Or is it something else I haven't thought of yet? <G> If anyone has any ideas, fixes, or workarounds please let me know. BTW, the diskette was produced by the IC3 client-disk producer; the point of all this is so I can create an image of my hard drive using IC3 and put it on my HP-UX box for safekeeping. Thanks in advance for any help! =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Jayson T. Garrett, Mississippi College Computer Center P.O....

...ling=pri_cpe context=pri1 ;channel => 25-39,41-55 channel => 1-15,17-31 group=2 signalling=pri_cpe context=pri2 channel =>32-46,48-62 group=3 signalling=pri_net context=pri3 channel => 63-77,79-93 group=4 signalling=pri_net context=pri4 channel => 94-108,110-124 -- TIA, Bruno at ic3s dot de

Hi all, I've programmed on a SNOM function key the destination "**77" which in my extension plan reprograms/toggles an asterisk DB variable which I use in the extension plan for some call routing decisions. I would like the SNOM extension light to permanently reflect the current toggle status of my application logic/asterisk DB variable. Is something like that possible to do in

Hi all, I' m currently testing a Wifi UT Starcom F1000 and for my taste even with the audio volume level even on max. setting, the speaker is not loud enogh. Is there a way to raise the audio-level specifically for a concrete SIP device via a sip.conf setting or something like that ?? TIA, Bruno

...ing a SIP - Target in the above Dial() statement the reached target is able to use *2. What is the current status of this feature in asterisk? What is needed to make it work with Zap-Channels ?? features.conf [featuremap] atxfer => *2 ; Attended transfer TIA, Bruno -- bruno@ic3s.de

...... it seems to work and then the job just up and vaporizes... ----- If the Creator had said, "Let there be light" in Ankh-Morpork, he'd have gotten no further because of all the people saying "What colour?" -- (Terry Pratchett, Men At Arms) Keith Baker kbaker3@ic3.ithaca.edu Ithaca College (607) 375-2602 Theatre Production Arts http://thelighthouse.dhs.org/keith/

do you mean it cant be circumvented from the perspective of syslinux or TPM? Sadly firmware exploits are rampant. Core Security has good research on this as phrack... SMM can be gotten to by far too much. Tho if you are targeted by that skill level then going to ic3 might be good idea. -------- Original message -------- From: Ian Bannerman <ian at internals.io> Date:04/29/2014 2:43 PM (GMT-05:00) To: Gene Cumm <gene.cumm at gmail.com>,"Taylor Jr, Matthew [U.S. Computer Corp]" <matthew.taylor at chevron.com> Cc: syslinux at zyt...

...ist to use SetTransferCapability(VIDEO) but this does not change the behavior. Is there a way to set or preserve the bearer capability for the outgoing call leg? cheerio Steve --- environment pbx-test*CLI> show version Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q built by root @ pbx-test.bb.ic3s.de on a i686 running Linux on 2006-06-16 10:17:00 UTC with A Quad ZAP Pri Card ---- dialplan ;exten => 297,1,SetTransferCapability(VIDEO) exten => 297,1,Noop() exten => 297,2,Dial(${TRUNK}/0175234567) exten => 297,3,Hangup exten => 297,104,SetVar(PRI_CAUSE=17) ; Indicat...

...unconfigured channel 255/255 span 5 The Modems connected to this span get NO DIALTONE for every ATD. Modems on other spans continue to operate. After restarting asterisk everything is okay again for a while. Any hints about what is going on here are greatly appreciated ;-) TIA, Bruno - bruno @ ic3s.de ------------ Connected to Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h currently running on pbx (pid = 20305) Verbosity is at least 5 pbx*CLI> show channels Channel (Context Extension Pri ) State Appl. Data Zap/25-1 (pri1 s 1 ) Up Bridged Cal...

Hi all, using latest asterisk-svn I want to reflect an video call incoming via an PRI EuroISDN channel to another outgoing PRI channel, and I want the the outgoing channel to have the exact same bearer capability < Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) < Ext: 1 Trans mode/rate:

...stalled. only don't know why am I getting that one: Warning: loading builtin ole32.dll, but native version already present. Expect trouble. why, ah? forcing dll-overrides from wine cmd-line brings more joy: [s2@localhost c]$ pico ~/.wine/config [s2@localhost c]$ cat ~/.wine/config | grep -iC3 ^[^\;].*ole32 [DllOverrides] ; some dlls you may want to change "oleaut32" = "builtin, native" "ole32" = "builtin, native" "commdlg" = "builtin, native" "comdlg32" = "builtin, native" "shell&quo...

Hello list. I'm trying to rebuild the 2.6.9.67.0.20.EL kernel, but it fails even without modifications. How did I try it? Created a (non-root) build environment (not a mock ) Installed the kernel.scr.rpm and did a rpmbuild -ba --target=`uname -m` kernel-2.6.spec 2> prep-err.log | tee prep-out.log The build failed at the end: Processing files: kernel-xenU-devel-2.6.9-67.0.20.EL Checking

Hi, how may I define in a callfile the CallerID presentation to be used for the requested call, eg. set it to prohibited? TIA, Bruno -------------- next part -------------- A non-text attachment was scrubbed... Name: Bruno.Voigt.vcf Type: text/x-vcard Size: 270 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050913/fcb5c595/Bruno.Voigt.vcf

Hi all, I'm looking for a way with any asterisk-version with TE410P (cpe EuroISDN, Q931) for sending an INFORMATION ELEMENT KeypadFacility, eg. *87, during a connected call to the PSTN switch. Are there existing functions in asterisk to generate & send such IE ? If not what existing modules would be best to derive from? TIA, Bruno -------------- next part -------------- A non-text

Hi, I'm running asterisk 1.2.1 and started to play with iaxmodem 0.0.7 running on the same box. I wonder how to setup the iax account correctly so that I may initiate outbound calls via iaxmodem? registration upon iaxmodem startup is okay and I can direct calls to it. -- Registered IAX2 'iaxmodem' (AUTHENTICATED) at 127.0.0.1:33874 But upon an outbound call setup request from

Please reply to my email address as I can't get that uch email a day or my schools account starts dropping mail... Could someone please help, I have a Dec Alpha running Linux with Samba. I can log onto it with net use, but I just can't get the pain in the butt to appear on browse lists. I think the first problem is that nmbd is not looking up anything. When I do an nmblookup it

Hi all, I'm using Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o on box A with a TE410P (EuroISDN cpe) connected to another similar asterisk box B acting as EuroISDN master. I'm performing some load tests by contiously feeding up to concurrent 30 call files to /var/spool/asterisk/outgoing/ on box A which inititate via a dialplan context/extension a outbound call (redirected via chan_local) to