similar to: Square Key system

> Message: 1 > Date: Fri, 21 Jan 2005 17:38:27 -0600 > From: "Henry Devito" <hdevito@qwest.net> > Subject: [Asterisk-Users] SPA-2000 > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: <000d01c50012$4ea49f30$4300000a@homeacxa7jw2xn> > Content-Type: text/plain; format=flowed;

Hello all. I just received a TE110P and am trying to hook it to my Mitel 200SX has anyone successfully done this? My configuration is as follows. Asterisk -> TE110P ->Kentrox (csu/dsu) -> Mitel T1 Card. All I get is a blinking yellow on my TE110P card and an alarm on my Mitel. T1 card. Any advice would be great. Zaptel.conf span=1,0,1,d4,ami e&m=1-23 dchan=24

Which sound file is the one you hear when you call voicemail and it says Comedian Mail? I can't find it in the sounds directory -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050329/a007d73a/attachment.htm

Before I go and try to write something myself, I'm curious if anyone has a script that they're using for setting and clearing the MWI on a legacy PBX. I need to pick up a Zap channel and dial #63XXX to set the MWI, or #64XXX to clear it, where XXX is the extension number. One complication is that I've got a couple dozen extensions to handle the MWI for, and only four channels to work

Check my reply to your last post. Use SetGroup and Checkgroup before sending the call to your agents. Robert Jackson -----Original Message----- From: Henry Devito [mailto:hdevito@qwest.net] Sent: Thursday, September 30, 2004 10:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Queue Setup almost got it Ok I think I have the queue

Hi, I am on my next venture now, Need to set up 3 queues. I would like these setup using the agentcallbacklogin. Does anyone have an example of this? I have looked through the wiki , but all that did was confuse me. One of the problems I'm having is how do I configure my extensions.conf to dial the agentcallbacklogin -------------- next part -------------- An HTML attachment was

I am hoping to replace my Nortel 8x24 switch with Asterisk. Right now my cabling comes from my outside phone box into my office and into a punchdown block and leaves the punchdown block as an amphenol connector which currently plugs into the Nortel swicth. A second amphenol connector them plugs into the switch and extends to another punchdown block that I believe carries the lines throughout the

Lots of interest here for conferencing. I've probably convinced more people to start using asterisk@home for this feature than anything else. Can I input some suggestions; Need to change the rinky dink call icons (let me know if you need some better samples) Need to change the ability for a muted caller to flag they have a question and a desire to talk (I do a lot of mute all except 1 or 2

Hi, I have not implemented any of the spa-2000's yet. Do they work ok with asterisk? Is the 2000 capable of having 2 FXS extensions off each one or is it two fxs ports with the same extension?

Has anyone had successfully installed more then two digium wildcards in the same machine? I'm going for four. thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

Hi, Has anyone had any luck using modems on ata's other than with Cisco ATA-188's? I really don't have the money pay for the 188's as this is for my personal use. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041003/5ffde3f4/attachment.htm

Hi all, I bought a couple phones for really cheap just for a simple solution. I'm trying to get a few 7910 to work with *. I'm just not sure how to get them to work. The 7910 just sits there "configuring IP" Here is a copy of my skinny.conf. the extensions.conf is default. I just want to bring the system up in default before a start making changes. Do I need to make

I've narrowed this down to a single kernel (kernel.ll), which does a fixed-size matrix-matrix multiply: # ~/llvm-32-final/bin/llc kernel.ll -o kernel32.s # ~/llvm-33-final/bin/llc kernel.ll -o kernel33.s # ~/llvm-32-final/bin/clang++ harness.cpp kernel32.s -o harness32 # ~/llvm-32-final/bin/clang++ harness.cpp kernel33.s -o harness33 # time ./harness32 real 0m0.584s user 0m0.581s sys 0m0.001s

Does asterisk support the intel 537/md3200 chipset? I don't want to start any flames here, I know all about using generic crap in asterisk,[*] which I really don't approve of other than for testing, but I have a customer demanding a generic chipset for his one backup analog line. He will not spend the money for a Digium card and says he will find another company if I can not provide a

I'm seeing a difference in how LLVM 3.3 and 3.2 emit unaligned vector loads on AVX. 3.3 is splitting up an unaligned vector load but in 3.2, it was emitted as a single instruction (details below). In a matrix-matrix inner-kernel, I see a ~25% decrease in performance, which seems to be due to this. Any ideas why this changed? Thanks! Zach LLVM Code: define <4 x double> @vstore(<4 x

Hi, Yes. On Sandybridge 256-bit loads/stores are double pumped. This means that they go in one after the other in two cycles. On Haswell the memory ports are wide enough to allow a 256bit memory operation in one cycle. So, on Sandybridge we split unaligned memory operations into two 128bit parts to allow them to execute in two separate ports. This is also what GCC and ICC do. It is very

I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had

Hi everyone, I'm a newbie to Asterisk. Will Asterisk run on RH9, easily or does it have to run on FreeBSD? Will the drivers for the Digium cards work on RH9? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040919/7d981e5e/attachment.htm

Nadav, We see multiple regressions after r172868 in ISPC compiler (based on LLVM optimizer). The regressions are due to spill/reloads, which are due to increase register pressure. This matches Zach's analysis. We've filed bug 17285 for this problem. Is there any possibility to avoid splitting in case of multiple loads going together? Dmitry. On Wed, Jul 10, 2013 at 1:12 PM, Zach

I set up the pilot number to voicemail to be 777. When a user calls 777 the voicemail answers and asks for mailbox, then password. Is there a way for the Voicemail to read what extension they are calling from and just ask for the password? I have a person complaining because they have to enter their mailbox number every time they check their voicemail and the "old" pbx didn't ask