search for: circuitous

Hi all. Is there an electronic circuit design CAD package available for Centos 5.5 please? Kind Regards, Keith Roberts ----------------------------------------------------------------- Websites: http://www.karsites.net http://www.php-debuggers.net http://www.raised-from-the-dead.org.uk All email addresses are challenge-response protected with TMDA [http://tmda.net]

Justin Holewinski <justin.holewinski at gmail.com> writes: > int globalIndexY2 = get_group_id(1)*186 + 6*get_local_id(1) + 2 + 1; > bool valid2       = validX && globalIndexY2 >= 4 && globalIndexY2 < 3910; > > Clang, even at -O0, is performing short-circuit evaluation of these > expressions, resulting in a fair number of branch instructions being >

A lot of people are going for the "VOIP only" approach, but SBC says you have to have an active analog voice circuit before they will sell you DSL. Does anybody know which DSL providers will sell you DSL without making you pay for a voice circuit? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com

Is there any way to disable short-circuit evaluation of expressions in Clang/LLVM? Let's say I have C code like the following: bool validX = get_group_id(0) > 32; int globalIndexY0 = get_group_id(1)*186 + 6*get_local_id(1) + 0 + 1; bool valid0 = validX && globalIndexY0 >= 4 && globalIndexY0 < 3910; int globalIndexY1 = get_group_id(1)*186 + 6*get_local_id(1) +

Hi- Perhaps someone with an E100P in hand can answer this: I just received an E100P from Digium (I normally buy quad boards) I noticed that the circuit board says "T100P" on it, and I assume that the T100P and E100P both use the same circuit board. Can someone please confirm that their E100P says "T100P" on the artwork? Thanks Scott Scott M. Stingel Emerging Voice

I have release my routines for PRI circuit monitoring. You, your client or anyone can be notified by phone, beeper, email or txtmsg that your circuit is down. If Asterisk crashes due to an oscillating circuit (as I have found it sometimes does), sendmail is usually intact and email notification and txt messages will usually get through. If the client has backup lines, and Asterisk remains up,

10.10.2011, 18:29, "David A. Greene" <greened at obbligato.org>: > Justin Holewinski <justin.holewinski at gmail.com> writes: > >>  int globalIndexY2 = get_group_id(1)*186 + 6*get_local_id(1) + 2 + 1; >>  bool valid2       = validX && globalIndexY2 >= 4 && globalIndexY2 < 3910; >> >>  Clang, even at -O0, is performing

I have 2 servers, configured identically. Each has a TE405P and 2 PRIs. One server was experiencing crackly audio on one circuit, accompanied by HDLC bad FCS messages. The telco recabled and moved me to another port on the DMS-100. The audio is better, but there are still bad FCS problems on the span. I have moved the PRI in question to the other server, and the problem does indeed move with the

Anyone know who that good looking female is thats on the Digium.com website ? Ok, my Real question is I noticed that Digium has relesed a new T1 card with an echo canceller. I also noticed that its supports E&M Circuits. Im I have very little knowledge on T1 circuits and traditional PBX's so what Im asking is can I use Digiums T1 card to connect to another PBX via a tie line ? Or does

https://bugzilla.samba.org/show_bug.cgi?id=2240 ------- Additional Comments From rse@engelschall.com 2005-01-13 07:48 ------- Created an attachment (id=887) --> (https://bugzilla.samba.org/attachment.cgi?id=887&action=view) rsync.patch.lastmatch -- Configure bugmail: https://bugzilla.samba.org/userprefs.cgi?tab=email ------- You are receiving this mail because: ------- You are the

Hi, We have quite a large setup working just fine most of the time. We have 60 outgoing lines on PRI and we never use all of these lines. But sometimes we get the "all circuits busy now" message, seemingly random. Sometimes we get it before the call even goes through to PSTN. Sometimes after 5 or 6 rings etc. It seems that the carrier is signalling something and asterisk always

Does anyone here know anything about GEN-GEN analogue circuits, also known as Manual Ring-Down (MRD)? Apparently they are widely used in Hoot'n'Holler systems for financial dealer-boards. I have been asked to try and interface to such circuits, and have been having great difficulty locating any specifications for the interface. Apparently, they are always-on 2-wire analogue circuits with

Hi, I am attempting to make about ten calls simultaneously and intermittently get 'SIP/voipprovider is circuit-busy' followed by 'everyone is busy/congested at this time" I am not sure if this is related to my bandwidth to my voip provider, a configuration issue or something else. Has anyone seen this before and have any suggestions. Thanks in advance. --------------

I was posed this question: A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Well, if you set that T1 up to carry data and run a link between two IP networks over it, how many SIP conversations could it be expected to carry? How about IAX? How would one extend this calculation to varying bandwidth circuits and various VOIP protocols (MGCP,

I've got a system with 5 pri circuits configured into a CFAS group with a primary and secondary d channel. There are three TE410P cards in the system. The 5 circuit span are located as follows: circuit 1 on span 5 circuit 2 on span 1 circuit 3 on span 6 circuit 4 on span 2 circuit 5 on span 9 primary d chan is on chan 24 of span 5 (chan 120) secondary d chan is on chan 24 of span 1 (chan

When using a PRI on my asterisk system, and when calling a busy number I often get an "All Circuits Are Busy" message. I've trouble shot it and made absolutely sure all circuits are not busy... but the number being called is in fact the issue. How do I get Asterisk to give me a busy signal instead of "All Circuits Are Busy" when calling a busy number? Is this something

Hi everybody After setting up trunk with FWD, all I get on my Asterisk box is message saying that all circuits are busy now, try your call later. Even 612 (time) says the same thing. Why is it that and how can I fix it. Zeeshan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, we've had an Asterisk solution working for quite awhile. Today our IAX2 carrier started reporting circuit-busy to all calls for approximately an hour, which is different than a busy condition. I'm guessing that this indicates the carrier was having an outage, or was handling too many concurrent IAX2 calls. Is there a way to change our dialplan to fail to PSTN in case Dial(*)

I call the firm from my portable at home (zoiper softphone). I have internal extension 60, and I call the internal SIP-client 10 at the firm via an IAX-connection over internet. My colleague at phone 10 answers my call. I ask him to transfer me with my colleague at extension 50. He then presses "transfer" on the grandstream GXP2020 (I get music) and dials the number 50. Phone 50

I am not sure why I am getting this message, I have an outbound route that goes to asterisk gateway1 then asterisk gateway2 When all lines on asterisk gateway1 are full, I get the message " all our circuits are busy now" then few second later, the phone rings, going to the second route! And the call can be established, how can I get rid of this message?? thanks --------------