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Subject: Heartbleed Toolkit | Secure, Detect, & Repair Date: Thu, 10 Apr 2014 18:12:16 -0400 From: Red Hat <email at engage.redhat.com> View in a Web Browser <http://app.engage.redhat.com/e/es.aspx?s=1795&e=352069&elq=852ad1748d834dbeac7f2adf6f4b1679> "Follow us on Twitter"

A concrete example: We''re building a system to organize the information about workshops being held in various conventions. A convention has more than one workshop, and each workshop can be held in more than one convention. Also each workshop has a host, who is specific to a workshop being held in a specific convention. For example, Matz may host an "Introduction to Ruby"

Hi All, I've been wondering if I can instruct asterisk in the dialplan to engage the Media handling for a particular call or not. I've SIP users behind Kamailio & RTPProxy, and I can make use of sip.conf setting "directmediadeny|directmediapermit" to offload media from asterisk for peer-to-peer calls BUT what if someone wants to record a call or engage some feature-code ?

I am slowly getting up to speed with asterisk. This is a very basic problem but I would appreciate any help. I am using a small network of clients and an asterisk server. Each client has a headset to communicate. Is there a simple way of playing an engaged tone when a line is busy? Regards ******************************************************************** This email and any attachments are

Hi, Is there a way to know if a member of a queue is currently engaged on a call? Or if a queue can return a busy status if all members are currently engaged in a call? QUEUESTATUS only returns FULL and TIMEOUT, and the scenario only falls into TIMEOUT, and has to finish the assigned number of seconds into the QUEUE CMD before it falls back to the next routine on the dialplan. Any ideas?

+1 to Justin's comment The only standard for revert should be: it's broken, and here's a reproducer.  Nothing else should matter. ... says the person w/a ton of internal regression testing which regularly finds crashes upstream, and is terrified of the implied effort of having to engage each author of a broken patch individually while being unable to ship or see what other

This is kind of what I was getting at with my original email, so thank you for wording it better than I did. If we can agree that "contact the author first for internal bots" is better than "revert automatically, even for internal bots" (which may not be the case, I don't want to speak for others), then the problem becomes one of defining what an "internal bot"

Hi, I'm using zaptel with three way calling and call transfers with a hookflash. If I try transfering a call to an extension that is engaged I get an engaged tone. This is fine, but how do I get back to the caller? If I hookflash again I seem to put on a three-way call and the caller can hear the beeping. I can press hookflash again but I'd prefer the caller to hear only the hold

> On Oct 28, 2016, at 1:59 AM, David Chisnall <David.Chisnall at cl.cam.ac.uk> wrote: > > On 28 Oct 2016, at 09:46, Mehdi Amini <mehdi.amini at apple.com> wrote: >> >>> On Oct 28, 2016, at 1:21 AM, David Chisnall via llvm-dev <llvm-dev at lists.llvm.org> wrote: >>> >>> On 28 Oct 2016, at 02:43, ジョウェットジェームス <b3i4zz1gu1 at

Hello, I'd like to be able to see if a channel is use and handle the call differently if it is. The best I can find is the command ChanIsAvail(). The problem is, I have an snom200 phone which does call waiting, so even if it is engaged in a call, a second channel is still available on it. I would like to be able to differentiate between these two cases: no calls engages, or calls

Hi I've got an 8 port Junghanns ISDN BRI card with 4 ISDN lines connected to it; all 8 channels are configured in my zaptel.conf. I am having a problem where intermittently, but very often, the ISDN line is engaged when the user tries to make a call. It almost seems like the previous calls are not being disconnected once the user hangs up. I no there is a priresetinterval setting in the

Hi All, One of our user asked the question, when she tries to call another local extension but the other end is engaged she will keep on trying until she finally can get thru. So she asked would it be possible to request for an auto-callback from the user she's trying to call to once it's not engaged anymore. is this possible on asterisk? what is that feature called? i am using

On 28 Oct 2016, at 10:10, Mehdi Amini <mehdi.amini at apple.com> wrote: > >> >> On Oct 28, 2016, at 1:59 AM, David Chisnall <David.Chisnall at cl.cam.ac.uk> wrote: >> >> On 28 Oct 2016, at 09:46, Mehdi Amini <mehdi.amini at apple.com> wrote: >>> >>>> On Oct 28, 2016, at 1:21 AM, David Chisnall via llvm-dev <llvm-dev at

Thanks for all the help on my previous histogram problem. I intend to summarize the solutions back to the list Real Soon Now, but first, I've got another problem. I've made a bar chart that reports means. I'd like to put the number of observations on top of each bar. Here's what I've got: barplot((subset$x), col=grey(.5), ylab="Mean Engagement

I have a functional pair of computers, one running a DC and the other running as a member server providing storage for the W7 clients. Within the member server, many directories have modified ACL settings to share among W7 client/ADDC users. I would like to backup the member server and protect the ACL and extend attr file permissions. Rsync will do this with "-A" and "-X"

Reid said: > I don't think whether a buildbot sends email should have anything to do > with whether we revert to green or not. Very often, developers commit > patches that cause regressions not caught by our buildbots. If the > regression is severe enough, then I think community members have the > right, and perhaps responsibility, to revert the change that caused it. > Our

I'm not sure if this is an Asterisk thing, a handset thing or a telco thing, so please be gentle with me if this is not the right place to ask ..... When placing a call over a SIP channel to a mobile phone, if the phone is engaged, it does not return a BUSY status straightaway. Rather, I get a ringing-out tone for the timeout duration specified in the Dial() statement; *then* I get

When trying to dial out 982420173 our main number I get the engaged signal before I finish entering the phone number Any ideas ???? Regards Mick

I am developing apps using ARI which need suppression of DTMF tones in the audio, and I have been told (back in December) that asterisk depends on SIP providers to suppress DTMF tones in the audio stream. Having sorted out my ARI code to suppress DTMF as I wanted, it turns out that SIP providers are not very good at doing that suppression (leaving audible clicks, or failing to suppress the tones

I'm trying to build a test suite so that I can run "calls" through and verify the call results. I've made a cross over cable and linked my 2 ISDN30 ports together. So now I can dial out on span 1 , and to receive the call on span 2. in the context for span 2, I have the following: <snip> ; #1 "answer" a call and play music 000XXX : ring for a random period,