similar to: Test

Hi Everyone, I'm planning my first asterisk box, and I'd like to know what SIP providers everyone likes. Voipjet? Gizmo? Somebody else? Thanks, Alex

Hey all, I've been trying to get MoH to work from the line-in on my soundcard, but as of yet have had no success. I found this script that should allow for it to happen: http://www.sineapps.com/news.php?rssid=722 The script, when run as the asterisk user, works properly and streams sound to stdin. But when Asterisk starts MoH it stops it immediately afterwards with no explanation. Has anyone

On Mon, Jun 12, 2017 at 09:51:53AM +0200, Reindl Harald via samba wrote: > Am 12.06.2017 um 09:41 schrieb Mike Brown via samba: >> On Mon, Jun 12, 2017 at 09:28:20AM +0200, Reindl Harald via samba wrote: >>> Am 12.06.2017 um 09:03 schrieb Mike Brown via samba: >>>> On Mon, Jun 12, 2017 at 01:53:10PM +1200, Andrew Bartlett via samba wrote: >>>>> On Sun,

I've inherited a large R codebase which has grown over a few years and a few different developers. It contains many things I'd like to delete: - Unused functions - Variable definitions that are never called - Unreachable code I'd write that myself, it would even be fun, but I don't want to reinvent the wheel. Is there an R package that can find these things? I've heard of

Dear List , We are using intel 82599EB based SRIOV capable 10GbE cards on the host machine.The VFs are being assigned to the guest in pass through mode and are providing the network connectivity as expected. However when we shutdown the guest the connectivity does not comes back via the VFs on the guest in next boot unless the VFs are detached from the *guest* *before* initiating a shutdown of

Does anyone have any suggestions on how to best represent an assumption statement(*) in IR? In particular, I want to expose the information implied by the assumption to the optimization passes without emitting code (after optimization) to check the assumption itself. I've tried a couple of options so far, and none have gotten me quite the right semantics. Has anyone else implemented

mvbutils::foodweb produces a graphical display of the hierarchy (or network or ...) of function calls. Isolated functions are not called. This might help you. -- Mike On Wed, Nov 15, 2017 at 12:44 AM, Alexander Engelhardt <alex at chaotic-neutral.de> wrote: > I've inherited a large R codebase which has grown over a few years and a few > different developers. > > It

This seems to indicate that the idea is a workable solution to your use case. We've still got some bikesheds to get through. (And I don't consider myself a good reviewer for Clang in this, so we need to identify who is.) As background to this extension idea, let me say that I prefer the notion that developers, particularly perf-aware developers like gamedevs, will want to handle PGO via

I''m using 3.0.2-3 and have gotten a matching user report from someone using 3.0.1. After powering off a VM, changing its MAC address and rebooting, it becomes unreachable. I made sure all ARP caches expired first before trying to contact the VM. Looking at the network traffic shows the ARP who-has query hanging indefinitely. "brctl showmacs xenbr0" also shows the old,

I''ve just gotten wxruby-swig to compile with wxWidgets 2.5 on Mac OS X. Thankfully, it mostly consists of getting rid of already depricated methods from the widget classes. I need to try with Linux and Win32, but it should be a smooth transition. Nick

Hi everyone, I am looking for advice about the design of a SIP-based intercom. I count on your help, as my current attempts are not fruitful (yet). This will be a pretty long message, so here's my fundamental question: Is there a way to interpret DTMF tones sent by the calee (not the caller) while a voice call is in progress? Here's the desired scenario: - there is a box with

On 09/23/13 11:54 PM, Chris Lattner wrote: > > On Sep 23, 2013, at 5:25 AM, Alex L <arphaman at gmail.com > <mailto:arphaman at gmail.com>> wrote: > >> Hi everyone! >> >> Today is the official "pencils down" day for GSoC and I wrote a >> report describing what results I've achieved since my last report in >> July: >>

Greetings, I just received some netweb-301 phones frm Seshu down in NJ. I cannot for the life of me get it to register with the asterisk server, nor upgrade the firmware to the latest (1.41) i'm still using 1.37. The packets are traversing the router, going into the other subnet, hitting the asterisk box, but not actually making it to asterisk. Nothing in the asterisk logs, but tcpdump

First question, is there a forum for asterisk@home specific questions? I've asked what must have been questions about asterisk@home here and gotten some indication they weren't welcome. Second, does anyone know what files need to be backed up? I don't need to back up the entire system since I can reinstall from the CD in fairly quick order, however, other than the files in

Hello list, I need a hand to find the best dialplan failover solution when using two SIP Trunks. My reasons to do failover are: a) one of the two providers could be unreachable b) both providers may be UP but one of them could return a SIP error message (maybe caused by DOWN E1s) Googling I found a few possible solutions: 1.

On Tue, Jun 14, 2016 at 10:36 AM, Lawrence, Peter <c_plawre at qca.qualcomm.com > wrote: > Daniel, > > What am I missing in the following chain of logic: > > > > As far as constant-prop, value-prop, range-prop, and general > property-propagation, > > > > 1. the compiler/optimizer **has** to get it right for if-then-else and > while-do or

Hi, I'm a pretty new Linux (Ubuntu) user, only a few weeks, and I few days ago I started having troubles with Wine. I posted this over at the Ubuntu Forums but haven't gotten a reply yet, so I thought I'd try here as well. I'll just quote my posts. First post: > First off I'm new to Linux, so sorry if you have to explain something twice to me. > > Basically my

Is anybody using Nortel IP Phone? I have (second hand) Nortel i2002 phone and when it boots I get: DHCP server unreachable F/W version: 0604D9C My setting: DHCP? [0-No, 1-Yes]: 1 DHCP: 0-Full, 1-Partial: 0 Can any body suggest how to troubleshoot it? -- #Joseph GPG KeyID: ED0E1FB7

Ok - I have not gotten anything since 6:03 am this morning. It's now 4:33 pm - has anyone gotten anything?

Hi all, I've been seeing some strangeness lately on my 10.2.1 server. It's gotten to the point that a few times each day, I see masses of SIP clients becoming unreachable. They're not all on the same network, and we don't see any calls drop. In a few seconds, they all come back. I don't think it's a connectivity issue because we don't drop calls, and the endpoints