similar to: Safecom SIP-300 Information?

I am having a bit of a problem and hope someone on here can help (if it has been covered already please feel free to point me in the right direction - I have searched and searched but found nothing!) Setup: Linux box : Debian Sarge using Samba 3.0.10-Debian windows Box : Windows 2003 SBS acting as ADS master. I ran through the setup instruction and can connect from the linux box to the windows

I am trying to setup a Debian Sarge box as a file/print server for a Win2k3 AD. The linux box can join the AD domain (using kinit administrator@EBUYER.SHE then net ads join) A wbinfo -u gives me "error retreive user list" A wbinfo -g just list the groups in the "BUILTIN" object on the 2k3 server A wbinfo --sequence gives me : BUILTIN 1 EBUYER 1 Now, as far as I can tell

I am looking for a supplier of Digium hardware in Canada. Any suggetions? Thanks, Adi

Mike, No they have not. Calls are failing again today. They have offered to refund my money but that does not solve the problem. My asterisk server is only 4 to 12 ms away from their "network". I have had VERY good luck with nufone.(40 to 45ms away) Only have 1 or 2% fail rate. Going to be calling txlink.net on Monday. Seems that LiveVoIP does not care about asterisk users. They like

Good Day list I am looking to buy a few Wooksung Video phones to try with my asterisk box.... http://www.wooksung.com/eng/html/pro/pro_001.html has anyone had any experience using these with asterisk? Thanks Ron

I just signed up with livevoip for 800 DID and have very choppy audio. From PSTN to my asterisk is ok but asterisk to PSTN is terrible. I am using IAX and was assigned to server iax01.nyc.*. I do not believe it is a bandwidth problem on my end and I have no problems using iax with gafachi. I opened a ticket with livevoip but no response yet. Would I be better off using sip with them? Is there

Anyone suffering an outage with them right now? I am getting the following from my box when I try to dial using them.... == No one is available to answer at this time W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050309/e1096325/attachment.htm

Anyone use livevoip? opinions? -- JD Austin Twin Geckos Technology Services LLC email: jd@twingeckos.com http://www.twingeckos.com phone/fax: 480.422.1250

So it looks like Livevoip went Bankrupt ------------------------------------------- There is a Federal Court Order in place and has been since Friday early a.m. ALL Suppliers are now under a Court Order that prevents them from terminating any and all services to LiveVoip LLC. If they take such any action they will be in direct violation of a U.S. Federal Court Order. If you have any questions

Hi Everyone, I have a DID number with livevoip, but I have been experiencing two problems that I can't seem to resolve. I am not sure if they are in any way related. I have other DIDs with iax sixtel but I do not have that problem. Livevoip seem to think that the problem might be with my configuration. Can someone help me figure out this problem please. 1) When an incoming call to my

Hello All, I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using Asterisk@Home 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. >From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Does anyone know how to resolve this problem? Thanks, Wiley

After our department went to using *, I've had several inquiries about doing VoIP for my entire organization (Small county). We have ~10 locations with various links in between (Mostly p2p T1s, some Frame (1.544mbps commit), some ISDN, some VPN over 768kbit internet) Right now we're using several NEC Electra Elite systems, and 2 Nortel Meridian systems. In one of the main locations we have

Anyone having problems with LiveVoIP lately? I am seeing failed outgoing calls. Calls that are being routed to wrong numbers. DID's that ring busy. For the pass 2 days I am unable to pass CID. Is anyone else have these problems? Can anyone recommend a Quality VoIP provider?

> You have been replied to - we do not use digital certs, we do not > reply when you have some sort of Spam blocker. This time I am > responding even though that is not policy. > It seems it is their policy not to answer. FYI info I tried to get an account with them a week ago. I did not get any information how to setup, just that they cashed my credit card. Several calls to them

I am looking for a good provider for IAX2/800 termination. I am currently using FreeWorldTel and wanted to use NuFone but it seems that both of them don't provide customer service. FreeWorld has terrible voice quality and NuFone never answers their phone or responds to messages. Thanks, Linn

What KVMs do you use? I personally like the ConnectPro.

Hi all, Before I setup an account with them, I'd like to hear other people's impression of LiveVoip. I'm considering using them for 800 numbers, and I'd like to feel comfortable that others here on the list have had good experiences with them. Thanks, sorry if this is the wrong list for this. :) Sena

Here is my configuration everything would seems be straight forward, but I can not register both asterisk with each other. Both asterisks have Static IP but they are behind firewall/router, so it means I have to use Register statement. I'm a bit confused about the register statement. How can they can register with each other when both firewalls are blocking port 4569? Do I have to open

I have several DIDs (working well) with LiveVoip and I just signed up for some outbound minutes. Unfortunately they did not send connection instructions. I tried: exten => _1NXXNXXXXXX,2,Dial(IAX2/userid:password@217.160.244.186/${EXTEN}|60|s) but I get the error Feb 19 15:14:09 WARNING[21453]: chan_iax2.c:5546 socket_read: Call rejected by 217.160.244.186: No authority found --

A few months ago, the friendly folks from liveVOIP went under. We had some discussion on how to limit our losses, and my recommendation was a chargeback, since "FTTP Services" -- their CC merchant -- wasn't affected by the bankruptcy, as far as we could tell. Today, I received this from my CC company: http://muware.com/asterisk/livevoip.pdf Anyone else got lucky?