similar to: Re: Generate Random Numbers in dialplan

On Fri, 2006-10-13 at 12:52:38 -0400, Jon Weisman <jweisman@ibell.net> wrote: > Hi All, Anyone know how to generate random numbers in the > dial plan? I've tried using the RAND function but it doesnt > work. Basically I need to generate a random 5 digit number > everytime a particular extension is dialed and then save that > into

Hey, I'm going thru logs, and I see some very common and interesting things that the hackers are looking for. In a whole bunch of scans, I've noticed that the first guess or two for sip accounts is usually a 10-digit number. I'm asking myself, why these numbers? Are they looking for a voip trunk? Or is it just like a serial number for the scan? What? Here's some examples:

https://bugzilla.netfilter.org/show_bug.cgi?id=1328 Bug ID: 1328 Summary: Please allow ipset add and del via the /proc/net/xt_ipset mechanism Product: ipset Version: unspecified Hardware: x86_64 OS: All Status: NEW Severity: enhancement Priority: P5 Component:

Hello! Oh, wise ones, ponder with me over two of the surprises that populate the universe! I have a phone, that I sometimes cannot reach, connected via pjsip. It can call other extensions just fine, it can call out over a trunk to my cell, all is well, but getting a call? Forget it most of the time. Here is all the config relevant to that phone: [murftest12] type=aor qualify_frequency=1992

On Wed, Oct 29, 2014 at 7:10 PM, sean darcy <seandarcy2 at gmail.com> wrote: > On 10/29/2014 08:06 PM, Matthew Jordan wrote: > >> On Wed, Oct 29, 2014 at 5:16 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> >>> Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13? >>> >>> >> codec_silk for Asterisk 12 will most

I see the following paragraph in the Asterisk trunk LICENSE file: "In addition, Asterisk implements two management/control protocols: the Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface (AGI). It is our belief that applications using these protocols to manage or control an Asterisk instance do not have to be licensed under the GPL or a compatible license, as we believe

On Fri, 2006-12-08 at 04:26 -0700, Douglas Garstang wrote: > Hi Steve. > > Thanks, but unfortunately, I can't be involved in that. We are > running Asterisk in a production environment and we're using > 1.2, not 1.4. I don't have the resources to work with 1.4. > Last time I filed a bug against 1.2 I got told off. >

Hi everyone, I am facing lots for problem with CDRs in 1.6 and above versions,its shows wrong records when I do transfer(caller side and calee side),callforward,call parking.Is the present CDRs in 1.6 is enough for Complete billing.?What I need to do to make it proper.Please help me on this. Thanks Nikhil

? Hello-- Boy, it's been a long time since I posted to the user mailing list! Pardon me, I've posted this to dev also, but I thought the general users should also be aware of this. I'd like to announce a proposal to the Asterisk Community, that I introduced at Astridevcon last month. It is a new API for playing sound files (mainly speech). A pdf describing the Proposal in some

I see that Asterisk distributes soundsets for English/English-AU/Spanish/French, and Russian. There is code for several other languages inside Asterisk; how does one obtain the other soundsets? Also, I noted that the source sound files don't seem to be publicly available for the sound sets that Asterisk distributes. I would assume that the source sounds are all 44khz (or more) cd quality

In putting together the SoundPack code, I am looking at the various language/locale specific code, and wondering how it all really stands... So, share with me, non-English speakers, what is your experience and impression? I heard a few comments during AstriDevCon, that some of the languages are not quite right; some said their language was understandable, but... Would anyone be willing to share

I'm not going to bore you with all the stuff I've done since November here. I put it, and some examples, in the file update1.txt in the git archive. To read it, do a git clone of https://github.com/WyoMurf/SaySentence.git I a nutshell, I've upgraded the SayScript grammar to handle expressions in the file names, upgraded the current en, fr, it, hu, and some others, to use the same

On 7/5/19 10:50 AM, Doug Lytle wrote: > On 7/4/19 6:40 PM, hw wrote: >> This has again, and for no reason, ceased to work again after >> restarting asterisk.  No matter what I try, I can't create a >> certificate asterisk >> would verify. > > Have you considered using LetsEncrypt for a valid certificate? > > Doug > > What would be the point

crashing. So, as a first step to solving **that** problem, make sure asterisk is compiled with debug flags, dumps another core file, and then you do the "gdb asterisk <corefilename>", and get a stack trace. That should give us some idea of what happened. > > I have a fairly simple Followme sequence in place to see how it works > before I get into the complex scenarios.

I was encouraged to post this notice on both asterisk-users and asterisk-dev; sorry if this is overkill, but it **is** applicable to both communities. Since the report is fairly large, has a pretty graph, and the whole bit, it was thought that posting on it a website, and letting you browse it would be better than sending hundreds a 50K message. http://www.asterisk.org/SpeedvsSizeExperiment

In the past little while, we've seen a wave of attacks on asterisk, via the provisioning. It goes something like this: A. scan for IP phones on the internet, either via spotting something on port 5060, or via the port 80 web interface for the phone. Or, use web sites that scan the internet, and classify the machines, to make your work shorter. B. Once you get into the web GUI,

hi, i have problem with pass-through faxing with this scenario hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen virtual) - linksys ATA i can fax to fax2mail on hylafax but after upgrade asterisk2 to 1.4 faxing is not working hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen virtual) - linksys ATA configuration is same do you hava any idea what is

Hi Steve, > .. I'll try to sort all this out, and then I'll attack this > problem. Hopefully, I get it all into svn before the next release of > 1.4...! Just wondering if any new CDR functionality made it into the 1.4.16.2 release? I have looked through the ChangeLog for the 1.4.15 and 1.4.16.2 releases but didn't spot anything to do with changes in CDR handling. I for one

Hello-- In the interest of Symmetry, does anyone else in the world see any need for a device like the IAXy (or the SIP ones from other manufacturers, like the ATA186), but one that presents an FXO interface instead, so it can be connected not to phones, but the PSTN? murf

Hello! It's me again. I began a fairly large modification to my CDR proposal some weeks ago, and finally yesterday and this morning got enough accomplished to allow a commit and some peer review. Check the docs out via " svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs " This is a directory; in it you will find: CDRfix2.rfc.doc CDRfix2.rfc.docx CDRfix2.rfc.pdf The docx