similar to: Shared Line Appearance

I am having a problem with the Vmail Web app and was hoping someone has some advice. I have been able to install vmail.cgi. I am able to bring up the vmail web page and login with out problems. However, I never see any vmails listed even when there are vmails present in the directory [root@linux-home INBOX]# ls -l total 40 -rwxrwxrwx 1 root root 255 Dec 2 14:45 msg0000.txt

I have a legacy pbx that I know nothing about. The PBX is a merlin legend system (now owned by avaya). I have found information on the list that I can use a 100D T1 card to bridge the PBX to an asterisk system. We are looking to head this route as well as replacing are AUDIX VM system with Asterisk. Has anyone done this that wouldn't mind sharing some of the merlin specific setup?

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I am stuck in a situation where I have two te110p's and really need a te210P. Anyone interested in a trade? They are in working condition guaranteed. If this is the wrong forum for this, I am sorry... I will take it off list. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird -

What and When is the next version of Asterisk? /Obelix

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Anyone have a 1650 running successfully in production mode with 2-4 PRI's? I want to make sure I don't have a motherboard compatibility problem before I buy one of these. We are going to be using a Digium TE210P to start off with and probably moving to the TE411P down the road aways? thanks, Sean -----BEGIN PGP SIGNATURE----- Version:

I'm having a brain fart.... What's the proper syntax for dialing out via a sip g/w (Mediatrix)? Been trying stuff similar to: exten => _6X.,1,Dial(SIP/3091@205.22.93.1/${EXTEN-1}) where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did even try the IP. Rich

I am trying to implement a follow-me script (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am having a brain fart as I haven't a clue where to get started with what to do with this. From my main menu, I want the extension 300 to execute the script as follows: exten => 300,1,dial(sip/200,20) exten => 300,2,playback(pls-wait-connect-call) exten =>

OK... maybe I got a little anxious and ran out and bought a Tyan GX28 with dual Opteron (dual core) processors. (It is a nice server ;) ) I did neglect to find out that you can not manually set the IRQ's on this motherboard. I am now stuck sharing an IRQ with the ethernet controller and no foreseeable end to my dilemma. I have a Digium TE210P and zttest consistently runs at 99.97% which

No, its up to them which backend they want to use. Sounds like they think that GCC is super quick compared to LLVM. Looks like another fud fart out of google to me. 2009/11/12 Jon McLachlan <mclachlan at apple.com>: > Any plans to make LLVM work with Google's new language, Go? > >

Does anyone know of any obscur reference for detecting an incoming fax. I currently have AMP running and everything else is working great. Installed the spandsp patches and software... using the default AMP extensions.conf, I start sending a fax, I hear it pick up and transfer to voicemail after 20s. Fax is set for system... Here is the detail from the extensions.conf [global] FAX_RX = system

On Nov 12, 2009, at 10:25 AM, Edward O'Callaghan wrote: > No, its up to them which backend they want to use. > Sounds like they think that GCC is super quick compared to LLVM. Looks > like another fud fart out of google to me. Edward, this is no place for comments like this. Evan > > 2009/11/12 Jon McLachlan <mclachlan at apple.com>: >> Any plans to make LLVM

Erik de Castro Lopo <mle+la at mega-nerd.com> wrote: >> -msse for SSE code, -msse2 for SSE2 code, -msse4.1 for SSE4.1 code > > Yes, that was it. Brain fart. These flags were not needed on x86_64. > > Erik But now all C code is compiled with -msse2 and it won't work on older CPUs. Isn't it better to compile only necessary files with this flag?

Yea, I figured it out...brain fart on my part. Got it working, but getting some sort of TCPSocket error now... still trying. :) [root@jabez etc]# darkice -c /usr/local/etc/darkice.cfg DarkIce 0.6 live audio streamer, http://darkice.sourceforge.net Copyright (c) 2000-2001, Tyrell Hungary, http://tyrell.hu Using config file: /usr/local/etc/darkice.cfg Using POSIX real-time scheduling, priority

I have an array with dimension of 5001 (rows) by 40 (columns). This array is a series of power output from a wind turbine in kW. The rows correspond to wind speed from 0 to 50 m/s in 0.01 m/s step increments. The columns correspond to the air density values from 0.90 to 1.30 kg/m3 in 0.01 kg/m3 step increments. I have a vector of wind speed and air density observations (typically 8760 records

On Nov 12, 2009, at 10:25 AM, Edward O'Callaghan wrote: > No, its up to them which backend they want to use. > Sounds like they think that GCC is super quick compared to LLVM. Looks > like another fud fart out of google to me. Actually, after chatting with Ian about it, it's more of a case of the FAQ being poorly worded than them being anti-LLVM. If you read it closely, it says

Any plans to make LLVM work with Google's new language, Go? http://www.technewsworld.com/story/Go-Go-Google-Programming-Language-68622.html?wlc=1257974768&wlc=1258041607&wlc=1258047741

On Fri, 2018-08-03 at 22:07 +0300, Michael S. Tsirkin wrote: > On Fri, Aug 03, 2018 at 10:58:36AM -0500, Benjamin Herrenschmidt wrote: > > On Fri, 2018-08-03 at 00:05 -0700, Christoph Hellwig wrote: > > > > 2- Make virtio use the DMA API with our custom platform-provided > > > > swiotlb callbacks when needed, that is when not using IOMMU *and* > > > >

I have a more complicated function I am trying to write, but I run in to a problem when I want to add something to the plot from more than one data set while simultaneously controlling the appearance of the additional layer. # Toy data: foo <- data.frame ( x = 1:4, y = 4:1 , membership = c( "A", "A", "B", "B" ) ) bar <- data.frame ( x = 1:4 + 1 , y

I set up hints and presence monitoring on some Polycom phones connected to an asterisk server with the expectation that the phones that are "watching" other extensions would be notified when the other extension sis ringing, in addition to the other statuses (on the phone, statuses set by the user on the phone, not registered, etc). I can see when the line is in use, and when it is

Has anyone successfully implemented concurrent appearance of the same PBX extension on multiple SIP phones? When using Cisco 7960s under call manager, you can have several phones with the same line appearance, but the first user to seize a line makes it inaccessible to other phones. Under SIP operation it seems as though this is not possible, but we don't see group ringing definable for