search for: livevoips

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 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

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

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

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 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?

Receiving calls from LiveVoip DIDs results in dropped DTMF digits. I'm using SIP, not IAX, and I've tried this without a dtmfmode and with dtmfmode in all the various permutations. Note that LiveVoip does not instruct us to put any dtmfmod statement in. The server is set to do ulaw and I've verified that it is doing so. LiveVoip originally suggested that I go from IAX to SIP to

For outbound only, I have traditionally recommended VoipJet. They just recently has a spat of issues that seem to have resolved though. I am still using them via their east coast server and it seems to work quite well again. Cost is around 1.3 cents minute I believe. Use IAX and g711 for best quality to VoipJet. Thanks, Wiley -----Original Message----- From:

I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or

Hello. I have an * server set up on a public IP. I have SIP clients at three different locations, all behind NATs. I have all the SIP users set up this way: [user1] type=friend username=user1 secret=password1 callerid="User 1"<101> host=dynamic qualify=yes context=outgoing All three SIP clients are configured to use STUN (using stun.fwdnet.net:3478). Furthermore, I have

...terisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread On Monday 27 June 2005 14:31, Michael Di Martino wrote: > If this list spent at least half the time on helping other asterisk > admins as it does on trivial things like LiveVoips bankruptcy it just > might be a great list. > As it stands now this list is kind of useless. Most request for > assistance with asterisk problems go unresolved of unanswered. Lists with this number of new members have a repetiveness of the same questions which people sometimes get tire...

This is probably a good time to point out that there is a good litmus test for all Voip providers. PRIOR to purchasing anything, send them an email and request the sales information. Ask about their servers or their policies or anything you can think of. How they respond will tell you a lot. If it takes forever, you can tell that they are either really busy, really indifferent, or something in

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

One would assume they have better things to do as they are quite busy. I think this is just a proactive measure meaning they say you cannot do it upfront so that in the event of a problem, it was predeclared. As to the rest of the TOS, I could be wrong but I got the impression that the owner of VoipJet speaks English as a second language due to some email exchanges. If that is the case, the TOS

I am having two problems. The first one is about half the time asterisk fails to read the DTMF tones. The second is with my 3 DID's some times it goes through and other times it does now. Right now it does nothing. Sometimes it rings for ever. With no out put on the asterisk console. They don't like to answer the phone or respond to email's is a timely matter. Anyone else having

<For someone that places outbound calls only, in a fairly low volume, is there a recommendation for which one would be <best for me? <I have had continual audio trouble with LiveVOIP, though other services <(FWD) work fine, so I'd want something that has good audio quality. I will toss in my $0.02 and say that I have had good luck with Voxee, simple setup, good quality, not so