similar to: packages.digium.com

On 11 Mar 2015, at 17:53, Matthew Jordan <mjordan at digium.com> wrote: > On Wed, Mar 11, 2015 at 10:28 AM, Steven Howes > <steve-lists at geekinter.net> wrote: >> Anyone know where it?s gone?.. Appears to have been down all day. > The hamsters should be running in their wheels again now. Cheers Matthew. Give them some food from me. Steve

On Wed, Mar 11, 2015 at 10:28 AM, Steven Howes <steve-lists at geekinter.net> wrote: > Anyone know where it?s gone?.. Appears to have been down all day. > The hamsters should be running in their wheels again now. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org

I will be out of the office starting Wed June 9th and returning Wed June 16th. Please contact Mary at mary at accessgate.net cell 407-267-1463 or Jonathan at jsnyder at accessgate.net cell 407-267-0056 or call our main number 888-227-9337.

Hi All, Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. Cheers Steve

Hi to all the busy-level / busy-limit setting in sip.conf is available for Asterisk 1.4.22 ? This is a piece of my sip.conf: [202] type=friend secret=202 host=dynamic ; This device registers with us username=202 ; Username to use when calling this device before registration limitonpeers = yes call-limit = 2 busy-level = 1 The directive busy-level is ignored.... I've also tried

Hi, Given the recent increase in SIP brute force attacks, I've had a little idea. The standard scripts that block after X attempts work well to prevent you actually being compromised, but once you've been 'found' then the attempts seem to keep coming for quite some time. Older versions of sipvicious don't appear to stop once you start sending un-reachables (or straight

Hello, using Asterisk 1.8.12.2 case : I call with my cellphone to our public telephone number Our receptionist answers the incoming call and does an attended transfer to my colleague ( A ) Colleague answers and the receptionist tells him that I am on the other side. Receptionist transfers the call and I am connected to my colleague ( B ) My question is about the CallerID that the

AXvoice server hacked. Here are few working accounts USE XLITE to make calls.... Registrar/Proxy magnum.axvoice.com:9060 Free Sample account.... username=xMaxwellSmartx secret=thanksapache username=woodsy type=friend secret=haramikuttasala username=wumingzi type=friend secret=kickyourass Enjoy! B.R BaBa Jigger -------------- next part -------------- An HTML attachment was scrubbed... URL:

The company making sublime text gets few thousands of dollars of notional loss :) On Oct 5, 2015 8:45 PM, "Steve Howes" <steve-lists at geekinter.net> wrote: > Wonder what happens when an entire mailing list tries to use that key?... > > On 05/10/15 15:28, Optical Phoenix wrote: > > ---------- Forwarded message ---------- > From: *Sublime HQ Pty Ltd* <sales at

Nick- Are you wanting to recreate the dialplan Echo() application in stasis? Why not just send the call to Echo() instead of Stasis()? On Fri, May 22, 2015 at 11:25 AM, Matthew Jordan <mjordan at digium.com> wrote: > On Fri, May 22, 2015 at 4:41 AM, Nick Awesome <jleed at me.com> wrote: > > Can anyone tell me how can I create echo test using ARI stasis > application?

Did you open a Jira issue for this yet? I can actually work on this this week. On Fri, Oct 16, 2015 at 9:44 AM, George Joseph <george.joseph at fairview5.com> wrote: > On Fri, Oct 16, 2015 at 4:00 AM, Bryant Zimmerman <BryantZ at zktech.com> > wrote: > >> Is there a way to limit the items returned by pjsip show [type] using like >> > > There isn't but

Does IPKALL still exist? I am after a free SIP trunk - who is still giving these away these days? As I noticed Stanaphone is no longer in business? Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -------------- next part --------------

Hi All, Has anyone passed the tests using Asterisk: http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html I presume the same rules apply for scaling and possibly have OpenSIPS/Kamailio on the front? Thanks. -- http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com

hi all, Can someone help me with a kind of howto build call center around asterisk with all the necessary features like CTI, call recordings, call spying, real time monitoring etc? I will be glad if it is an open source code. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

From: Matthew Jordan <mjordan at digium.com> > > > >> If the INVITE request is not shown in the CLI with 'pjsip set logger > >> on', then Asterisk is not actually receiving the request. > >> > >> Does a pcap show the message being sent to the correct IP/port? If you > >> change the transports to bind to port 5060, does that change

I'm in the process of specifying the hardware for some new Asterisk systems which will be running a substantial number of conferences with recording. I was wondering what there is to choose between SCSI, SAS and SATA disks, in terms of performance for this kind of application. I will be using dual drives with kernel-based software RAID1. Any advice from experience would be appreciated!

Hi All; Any one used Digium IP Phones D40? I need to know if they are stable with good voice quality? Comparing to Polycom 330, which is better? Let us talk frankly although I know that we have to support Digium. Regards Bilal

Well, it breaks audio for all NAT endpoints, how can I fix this? > On 18 Mar 2015, at 15:48, Matthew Jordan <mjordan at digium.com> wrote: > > On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed at me.com <mailto:jleed at me.com>> wrote: >> Hey guys, >> >> have issues with reinvite, no matter what endpoint is calling asterisk >> always tries

Hi Everyone, I have a weired situation where calls in and out are proceessed all right but when I dial *97 Asterisk is literally choking when it comes to announcements like "Password" or "Call from 205-456-2222". Each one of those announcements can take like 10+ seconds to finish with most of it not even compoundable. I run "top" and there is no heavy load on CPU or

Can anyone tell me how can I create echo test using ARI stasis application?