similar to: Hard drive write cache

Disclaimer: Not a troll I'm curious as to this obsession with uptime is. All of the posts of this type are along the lines of "After X days, Y thing does not work but if I reload or reboot, it's OK" - so why not cron a reboot? Is it considered bad form or something like that? I reboot every night whether it is needed or not, not afraid to admit it, and everything works fine for

I have some general questions about marketing. Lot's of technical info but I was wondering how people are getting the business to begin with. I'm from the IT end of things but Telco is quite a bit different. Is cold calling still the way to go or networking? General stuff like that. Are there any resources on the web I can search for? Any suggestions would be appreciated.

Hi there, I am getting ready to set up a production Asterisk system. It needs to be stable. Upgrading, patching, rebooting, troubleshooting etc. are pretty much NOT an option once this thing is deployed. Like any phone system, it is expected to just work. Having said that, which is the best version and subversion of Asterisk to use? I was leaning towards 1.2 but it appears there are some

Hey guys, appreciate the input. Here are some thoughts. ADSI phones are out of the question. This is a business environment, I can't worry about my employees not knowing how to forward calls, answer calls when away from the multiline phone, and no ADSI phone will handle multiple lines that I have found. I would love to put 6 X100P cards in a case and run asterisk on it. but... I

Hello, I have an asterisk server running with 23 g.729 licenses. I have also purchased a sound file from thevoice.digium.com. I need to covert this file (uLaw, PCM I think) to g.711, g.729 & g.723 for use with an IVR system. Is there a way I can convert the files using the g.729 digium codec? sox? Thanks -Matt -- Matthew S. Crocker Vice President Crocker Communications,

Hello, I?m using virt-install to build a guest system with CentOS. The system boots up, times out with dracut-initqueue timeout and drops me into an emergency shell. If I exit the shell the install continues and I get a working machine. Any ideas? virt-install \ -n TEST \ -r 8192 \ --os-type=linux \ --disk=/vm-images/test.img,device=disk,bus=virtio,size=100,format=raw \

Hello, I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd like to be able to direct an inbound fax call into my TNT, have it answer the fax and send the image file over to Asterisk, or some other system to deliver to an e-mail address(s). I'm not sure if I need Asterisk to any of the call control or not. I'd also like to setup a print queue and have outbound

Hello, I'm trying to upgrade an Asterisk 1.2 linux box to Asterisk 1.4. I installed the following -rw-r--r-- 1 root root 10908541 Sep 21 13:25 asterisk-1.4.0- beta2.tar.gz -rw-r--r-- 1 root root 993921 Sep 21 13:25 asterisk-addons-1.4.0- beta1.tar.gz -rw-r--r-- 1 root root 80019 Sep 21 13:25 libpri-1.4.0-beta1.tar.gz -rw-r--r-- 1 root root 1523413 Sep 21 13:25

Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Does Asterisk BE come with a GUI management console for managing phones, queues, VM and the like? -Matt -- Matthew S. Crocker Vice President Crocker

So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry

Hi there, Here is my dilema. I have a new small business customer that wants me to put in a VoIP phone system for them. Based on their requirements, I have determined that it needs to be a "set it and forget it" type of thing like a lot of small business proprietary systems. At the same time they would like to be able to do minor dial plan changes themselves so I have determine

I''ve been trying to match multiple public IPs in an ingress qdisc. The idea is to allow these specific IPs and aggregate value of 256 kbits incoming to the interface. Can anyone tell me how this can be effectively done if at all possible with tc? Thanks in advance -- Corey Rogers <jrog@sunbeach.net> _______________________________________________ LARTC mailing list /

i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ -------------- next part --------------

Hi I can provide my customers one or several phone lines by using an ATA through the SIP protocol. Is there a similar box that would allow me to provide a PRI (23B +D) to a customer using SIP ? Thanks Patrick

Hi, I have a project where I have to backup (copy /dev/hda on a server) and recover (copy back) hardrives from the clients. I tried this with rsync using: BACKUP: rsync /dev/hda user@host:/path/to/image RECOVER: rsync user@host:/path/to/image /dev/hda There is a problem when accessing devices like /dev/hda. The transfer aborts after a few seconds with the error message: write failed on

Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110621/764a6fa9/attachment.htm>

Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system to work with *? I'm not getting inbound or outbound calls to work. (Inbound caller gets line busy tone.) SETTINGS FROM MITEL: I built a Crossover cable and connected it like this: PSTN--T1--ASTRISK--T1--OLD MITEL -Crossover Cable Pin-out: 1 - 4 2 - 5

Could someone explain to a non-US dummy the following phrases I have seen on the list. "I can provide you with tier 1 termination 6/6. I can blend or NPANXX breakout." "We provide US48 termination, blended rate for 1 MOU and above is .008 with 6/6." What is 6/6? What is US48? What is blended? What is MOU? What is NPANXX breakout? -------------- next part --------------

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

I have a small government department that wants me to implement a Asterisk installation, however, they connect to the Government PBX, a Mitel SX200, and want to keep the ability to do that. I know there is no chance to connect the digital extension lines, but would it be possible to have the pbx admins send analogue extensions over and have those lines interface through an FXO interface? Or