similar to: Asterisk in the NW

For those of you that are in the Portland, Oregon area: I am giving a talk today on Asterisk at the PLUG Advanced Topics Meeting. Details below. JT >From: "Zot O'Connor" <zot@whiteknighthackers.com> >To: PLUG LIST <plug@lists.pdxlinux.org>, > PLUG Announcement List <plug-announce@pdxLinux.org> >Organization: White Knight Hacklers >Subject:

Hello all, I am a relative asterisk noob so please bear with me if my questions are obvious. What I'm trying to do is get our analog PBX (A Merlin Legend) connected to VoIP. From all my googling and reading voip-info.org (and this list) it seems very possible. I just wanted to describe my setup and see if I'm going in the right direction. What I'd like to do is set up an asterisk box

Ottawa, Canada ? February 15, 2006 - PIKA Technologies Inc. today announced that they have integrated PIKA?s high-density analog computer plug-in boards with the open source Asterisk PBX, with the introduction of PIKA Connect for Asterisk. PIKA Connect for Asterisk is a software layer, available free of charge and distributed under the GNU Public License (GPL), which allows interoperability

Good day! so I've inherited an Asterisk PBX setup and it's busting my ass. Since the setup wasn't document and is not running on our standard operating environment I have to say it is an ad-hoc / experiment which has become critical infrastructure for the company and is now considered a risk. Because of this there is a need to rebuild and (this time) properly document the

I hadn't seen anything on the asterisk list but just in case anyone is interest. Regards, Dean Collins Cognation Pty Ltd dean at cognation.net +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). > -----Original Message----- > From: Murat Aktihanoglu [mailto:murat at unype.com] > Sent: Wednesday, 22 October 2008 9:57 PM > To: Dean Collins > Subject: Fwd: [wwwac] Thursday 23

Hey folks, Is anyone working on Asterisk (or other) presentation proposals for OSCON 2008 in Portland, OR? Here's the link, in case: http://en.oreilly.com/oscon2008/public/cfp/13 I'd love to see more Asterisk content there! Martin Smith, Systems Developer martins at bebr.ufl.edu Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221

NAT endpoint calling local endpount - switching to native_rtp then no audio, both of them have direct_media=no, Verbose log: -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in new stack -- Launched AGI Script /pbx/agi.php -- AGI Script Executing Application: (Dial) Options: (PJSIP/99/sip:99 at 192.168.1.73:5060,20) -- Called

Hi! I have problem with MEMDISK, because when I load modules in NW Client32 using NIOS.EXE commands, computer is restarted :-( Not everytime, but manytimes. The problem is with MSDOS 5.0, 6.22 and FreeDOS too. But when I create in DOS another RAMDISK and copy all data from MEMDISK to new RAMDISK and change settings and PATH to new ramdisk, all work fine. Directly from floppy or from

I have had problems like this before. Probably there is something important that I don't know about routing. Let me introduce to you "Lasso2", a CentOS 4 www server that has been working perfectly well for years. Now I added a second nw card (eth1), automatically using kudzu. I cannot get this dual nw setup to work. The first nw card (eth0) stopped at once working properly,

I am proposing this patch for allowing the network bridging setup script to choose an alternative network interface if it is determined that the root is mounted using NFS, which is assumed to be done over the NIC serving the default gw. In this case the prefix of the device is used to find other devices, i.e., if eth0 is the default device an alternative device may be an other ones starting with

I am designing a solution for a hotel booking call center with the following (mandatory) design: After the call from the customer with the booking agent is complete (and the Hotel PBX disconnects from the call), a second PBX takes over to conduct a survey of how the call went. Both PBX's are Asterisk based. So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects, the

For those of you going to LinuxFest, I''ll be speaking today at 10:00 in G-106. And if you miss the talk but see me wandering around the campus later, don''t hesitate to flag me down to say "Hi". -Tom -- Tom Eastep \ Nothing is foolproof to a sufficiently talented fool Shoreline, \ http://shorewall.net Washington USA \ teastep@shorewall.net

Hello, we are using Asterisk with Adhearsion as our application server, with another Asterisk box acting as the office PBX, where all office phones are registered. A REFER to transfer calls within the office results in the Adhearsion application call being dropped, because the leg between the PBX and the app server is terminated by the PBX following the REFER. Is there a way to configure

I'm trying to develop a systematic way of placing legends on (among other things) scatterplots of grouped data. The journals in my field can be very fussy about the placement of figure legends. Typically they want them to be in either of two places - the upper left corner of the plotting area, not obscuring any data points, or outside the plotting area in the lower right hand corner of

Given the data structure below and the call to ggplot2, how can I increase the size of the axis scale points, the line weight, and the size of the legend? ddata <-structure(list(year = structure(c(1L, 1L, 1L, 1L, 1L, 1L, 1L, 2L, 2L, 2L, 2L, 2L, 2L, 2L), .Label = c("2003", "2007"), class = "factor"), area = structure(c(7L, 6L, 1L, 2L, 3L, 4L, 5L, 7L, 6L, 1L,

Hello when using Asterisk version 13.12.2 I notice that it takes up to 30 seconds (sometimes even longer) for a call queue to call its members. Example 1 : [Nov 21 08:17:57] pbx.c: Executing [queue at pbx-routing:15] Queue("SIP/incoming-00000246", "myqueue1,,,,300,,,") in new stack [Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class 'default', on

"Tyler, Ross E" wrote: > configure determines that my HP-UX 11.00 box has pread (and pwrite) because the linker resolves them to symbols in my libc. > > however, i don't believe that this system call is implemented in my version of the kernel because smbd dies with a SIGSYS (bad arg > to sys call) in libc:pread. > > my workaround was to override what configure

Hi! I've got a Comdial PBX that I would dearly love to replace with an Asterisk box. However, for various reasons, it appears not to be in the cards. Regardless of what management does, or does not, want, our current VM solution -- some Dialogic card with a "KeyVoice" application -- is dying. I'm 90% sure it's hardware. I'd rather shoot myself than replace the

On Thu, Mar 19, 2015 at 1:47 AM, Nick Awesome <jleed at me.com> wrote: > NAT endpoint calling local endpount - switching to native_rtp then no audio, > both of them have direct_media=no, Verbose log: > > -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in > new stack > -- Launched AGI Script /pbx/agi.php > -- AGI

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