similar to: X100P dead?

I have read thru what other users have tried in this list when they have experienced seemingly similar issues to what I have, without success. I suspect there might be an issue regarding both the X100P and TDM04b cards being used in an Intel SE7525GP2 motherboard, as I had to even wait for a BIOS update from Intel in order to utilize my 3ware 9500 SATA raid controller on it (3ware Kb -->

I don't mean to be a troll in any way shape or form. I was on IRC last night and I observed the following convo. below. What do you guys make of it ? [02:14] <bkw__> Let me tell you how chidlish digium and Mark Spencer is. I walk into a restaurant with them all here at Astricon wearing my sangoma shirt and he asked me to leave. [02:15] <Dovid> u serious ? [02:15] *** mog

Hi All, Is it correct to say that by design, asterisk wont make use of any cards hardware dsp capabilities ? I don't think that any of the hardware cards currently supported have any dsp capabilities, but I wanted to know if for example, in the future a driver was written for a card that did have dsp capabilities, would asterisk be able to make any use of it ? I am only just starting out

Hello , I'm a student in Shahed university in IRAN . Now I'm become a member of our uni IT-Center . We are going to have an organized network with a en-bloc user authentication and proxy server. In comparison between MS AD and Samba , i want to choose Samba server ( cause of my belief in Open Source ) . Our plan have a Forest(root : shahed.ac.ir) and trees for faculties (like :

(REPOST, sorry if you get this more than once.) Hello all, (Not sure if this is more appropriate for user or dev list) Does asterisk have any sort of "standards based" api that can enable an application to do call control on the switch ? For example, if I am developing a call center application using asterisk, I would like to be notified of inbound calls and then be able to route

Hi All, I am wondering if I will be breaking the GPL, if I write for example, a channel driver or make some modifications to the astrisk source code, to interface at RUN TIME, through sockets, with a proprietary system. Eg. 1. I write chan_xxx + modify asterisk source (make changes + new code publicly available) 2. chan_xxx supports hardware by XXX Corp. 3, XXX Corps interface is

Hello All, I am new to *, and this is my first post on the user list. I have had success with making / receiving calls to a SIP hardware Phone and the Console Channel Driver. My SIP phone *requires* that I register with a SIP server. For this, I used the NIST sip presence server (a version that I downloaded almost a year ago). I have a problem with getting asterisk to register properly,

I just installed a new TDM400P with one FXO interface in slot 4 (how it came from Digium). This box is running Debian with a 2.6.10-rc2-mm3 kernel. After the make linux26 and make install in /usr/local/src/zaptel, I can see contents in /dev/zap but any attemp to touch for example /dev/zap/ctl gets a no such device or address ... Any suggestions?

On 08/28/2013 10:08 AM, Sebastian Pop wrote: > Hi,tic > > in lib/Analysis/RegionPass.cpp there are 3 occurrences of: > CurrentRegion->getNameStr() > These are slowing down compile times with polly. > I would suggest to either remove these calls, > or only turn on when the programmer asks for -debug. > > The reason for the slow pretty printing of types is the same as

Did anybody already attempt to strip down an asterisk config to an absolute minimum for a specific use? Let's say I have a home installation and want to use capi and iax exclusively, and load only the channels, apps, codecs, file formats I really need. Obviously, to dig through the whole stuff, while maybe being educational, is still a major task. I'm aware of the fact that requirements

Hi everyone, I have used used scriptalicious for the autocomplete. It works but with one problem. When scrolling down the list, the values not visible are not scrolled into focus. eg I use the keyboard down or up key. Here is my CSS <style type="text/css"> div.auto_complete { height: 100px; width: 350px; background:

Hi, in lib/Analysis/RegionPass.cpp there are 3 occurrences of: CurrentRegion->getNameStr() These are slowing down compile times with polly. I would suggest to either remove these calls, or only turn on when the programmer asks for -debug. The reason for the slow pretty printing of types is the same as previously discussed in: http://lists.cs.uiuc.edu/pipermail/llvmdev/2013-July/063755.html

I have been trying to write a selector function for Autocompleter.Local that will show a predefined Array and will scroll automatically to the first match. I have had success with with getting my list to populate and show all of the choices, however I cannot get it to scroll to the first match. Since I build the JS Array I can know exactly how many items are in the list going into my selector,

Hi All, I am currently using script.aculo.us and Autocompleter for a project which I am dealing now. I faced a situation for which I couldnt find any solution and I could not see any reference regarding this issue in the enhancement/bug lists of script.aculo.us either. Autocompleter component is working perfectly, if the information returned from the server does not exceed ~1000 LIs.

I've read quite a bit in the older mailing list posts and the wiki but I'm missing some simple point. 1) What is required to send an SMS to a mobile outside the office given: Channel: ZAP/1 send it to $SMS_RECIPIENT (which includes the final "extra" digit) via $SMS_CENTER=the national message center server for sending messages $MESSAGE= the message text How is the .call file

Hi David, Thanks for your efforts here. I have a few comments on your patch, although I realise it's still a work in progress. +class ConstantPool { + MCSymbol *Label; + typedef std::vector<const MCExpr*> EntryVecTy; Use a SmallVector here? + MCSymbol *getLabel() {return Label;} + size_t getNumEntries() {return Entries.size();} + const MCExpr *getEntry(size_t Num) {return

Moving discussion to llvm-commits now that I have a more developed implementation: http://lists.cs.uiuc.edu/pipermail/llvm-commits/Week-of-Mon-20131111/195401. html > -----Original Message----- > From: llvmdev-bounces at cs.uiuc.edu [mailto:llvmdev-bounces at cs.uiuc.edu] On > Behalf Of David Peixotto > Sent: Tuesday, November 12, 2013 11:09 AM > To: 'Amara Emerson' >

sorry to bother with this trivial issue, but i am loosing all my hair ;-) got 2 x100p's and * on a slakware box x-lite to x-lite works fine! i also have: #ztcfg -vvv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. and in extensions.conf i got: [locals] exten

Hi all! I'm new to asterisk and I'm trying a simple config with: - Debian GNU/Linux (unstable) - last version of Asterisk - a X100P card I have a problem with dial out from a SIP software phone (XLITE) to a public number (ex. my mobile phone), asterisk start the call, but nothing happen... If I run "ztmonitor 1" I can see the right RX level and if I try to make a call with an

Hi, I have got one X100P telephone card in my Asterisk server and it's working well. I have two phones lines and would like to install a second card so I can use both lines. I have installed the card and tried to set it up, but all to no avail ! Could someone outline the changes that I need to make (and in which .conf files) in order to get the second card going ? Thanks in advance,