similar to: voip-info.org mirrors?

Is it wise to use an outage to promote your business, not on the user's list and not multiple times? Put it in your signature or something ;-) Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Shane Breen > Sent: Wednesday,

Yes Dave Bour Desktop Solution Center 905.381.0077 dcbour@desktopsolutioncenter.ca For those who just want it to work... Giving you complete IT peace of mind. (Sent via Blackberry - hence message may be shorter than my usual verbose responses) PIN 4cc364db (as of March 24, 2007) ----- Original Message ----- From: asterisk-users-bounces@lists.digium.com

For the last few days I can not resolve voip-info.org from many DNS servers. It does resolve with some DNS servers but I suspect it may be related more to caching. Using the host command: host -a voip-info.org 130.179.16.23 Trying "voip-info.org" Using domain server: Name: 130.179.16.23 Address: 130.179.16.23#53 Aliases: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR,

The DNS for www.voip-info.org seems to be non-responsive. Is there a mirror of this invaluable resource site? Tx, Steve dig www.voip-info.org ;; Got SERVFAIL reply from xxx.xxx.xxx.xxx, trying next server ; <<>> DiG 9.4.1-P1 <<>> www.voip-info.org ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: SERVFAIL, id: 61402 ;;

Today the front page of http://www.voip-info.org/ was taken out by a spammer. It also seem the history page for http://www.voip-info.org/ was also nuked. I've restored the best I could using google cache, but still missing some information. Who is an admin on http://www.voip-info.org/ and can fix it? PB

Anyone has an idea what happend to voip-info? it stopped working about 24 hours ago. Nir S -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070314/23cdc0f6/attachment.htm

Hi, It seems BRI signalling settings are missing from http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf I would like to add those parameters : bri_cpe_ptmp bri_cpe bri_net Is this http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conftied to a specific Asterisk version ? Can I edit this ? Regards -------------- next part -------------- An HTML attachment was

Hi, If it is really true that the voip-info.org website is hosted on a DSL connection without static ip, I have a server in managed.com datacenter that can host it. I still have some ip's free, so tell me if you want to use it. Bandwidth will be on my cost the first terabyte every month. Server has plenty of space left on the HD. I offer this for free, heck, I even offer mail domain with it!

Hi I can't find out how to add more links to the Navigation box in voip-info.org. Can someone help me out, please? roy

Looks like the site is back up. Don't all hit it at once, it might go down again ;-) Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070315/11c4a2bc/attachment.htm

Anyone have any insight as to why voip-info.org has been up and down all day, and more importantly unreliable for the last month? I assume the bandwidth is being donated or something, but surely someone would be willing to donate reliable bandwidth as the knowledge hosted on the site (which is also donated!) is worth way more than the bandwidth. There is no doubt it is the best

Chan Chung Hang Christopher <christopher.chan at bradbury.edu.hk> wrote: >> Ah, well, if you want to keep the landlines, then yeah, I guess asterisk is the way to go. If your goal is to replace keyline systems, then asterisk definitely has that kind of support which, it appears, even Cisco's solution does not (from the mouth of Datacraft Asia personnel selling the school

Hi: I am looking into getting incoming IAX DID channels for our office. I've found a provider. What I want, though, is an incoming hunt group -- that is, say we have three lines: 555 1212 555 1213 555 1214 Calls coming in on 555 1212 may end up on any one of the three. If 555 1212 is busy, the call forwards to 555 1213, and so on. I was under the impression that this has to be done by the

A short status update: Yesterday 3 of the 4 disk drives in the RAID array on the server that hosts voip-info.org failed. The coloprovider is currently working to replace the drives and I'm hoping that the site returns to service soon. Tomorrow is looking most likely. I'd like to thank all those that have called or emailed to offer help and/or encouragement. I will definately be looking

Does anyone have a mirror of this running? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050715/5fe78cd2/attachment.htm

Hi, folks: I need to be able to have a single DID ring multiple remote (IP and PSTN) extensions, and then pass the call to whichever picks up first. I'm sure this is old hat -- lots of providers offer it. I see that Trixbox will do it, but it's not clear how it's doing it. They use different terminology -- a "ring group" and "hunt strategy" How can it be done

Finally there is a place for us to contribute doc & info about * .... BUT it is realy beginning to feel the heat of its popularity its takes me forever to edit pages on the site .... On IRC there are a number of service providers willing to donate CPU cycles & Bandwidth to support this resource ... Does any one know who owns the site, and how we can help him move to one of the generous

On Sat, 2007-07-07 at 08:39 -0500, asterisk-users-request at lists.digium.com wrote: > Date: Fri, 06 Jul 2007 12:02:53 -0600 > From: Stephen Bosch <posting at vodacomm.ca> > Subject: Re: [asterisk-users] Mushtaq Ahmed is out of the office. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID:

I listened to all the demos you showed. My ear discerns a little muffling and minor "slushiness" in the GSM files you sent, along with a much more narrow bandwidth, mainly on the high end side, and Allison either has a mild whistling s or slushy s sound in her voice or the producer didn't properly compress it to "de-ess" the recording. Or, I could just be rather tired.

Hi: Say I want to build an IVR application which sends an SMS message to a mobile telephone when the caller responds to a prompt in certain way. I think I can manage the part about generating the message and building something to actually send it. The part I'm foggy about is: how would I actually get the SMS message to the carrier? Discussions with the carrier have led absolutely nowhere