search for: livevoip

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...

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...

...ain 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 to pass the blame. -Tim On Sun, 2005-03-06 at 17:04, Mike Dent wrote: > Hmmm, I was contemplating going with livevoip, glad I read your post. > I'd be interested if they resolved your issues? > > thanks > Mike > > > &g...

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...

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: ht...

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?

...ir 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 put me just on hold "Our phone lines are REALLY busy" --- What a BS !!! > > a. You are NOT a LiveVoip customer. > b. We declined your credit card , no charge was ever made. > c. Never threaten this company - we will not tolerate it. > d. Do not re-apply for service we do not want your business. > > LiveVoip LLC Unfortunately I heard here on the list, that their service is so bad A...

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...

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/live...

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 d...

...ality to VoipJet. Thanks, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of John Goerzen Sent: Monday, June 27, 2005 9:29 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt] On 2005-06-27, Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> wrote: > http://www.nufone.net. I've been using them for the past 18 months with zero > technical hassle. Jerjer and Shido6 hang out on IRC. Nufone is not a "hand > holding" VOIP provid...

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...

...ions, 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 LiveVOIP set up like this for incoming/outgoing calls: [livevoip] type=friend username=peer1 fromuser=peer1 secret=password host=livevoip.ip context=from-livevoip Calls between location 1 (using a Sipura SPA-3000) and location 2 (using a Grandstream Budgetone 101) go through perfectly. Calls from the PSTN...

...eed to be fully indexed. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of steve szmidt Sent: Monday, June 27, 2005 3:46 PM To: Asterisk 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...

...e----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk@txpe.net Sent: Monday, June 27, 2005 8:47 AM To: asfhasterisk@qwest.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt] I was charge $25 for a toll free port. I don't know what the fee is for a local port, though I have one in progress. I haven't been able to find these fees listed on the website (yet). Maybe someone from Teliax can fill in what extra fees may be charged? Doug At 10:17...

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

...heers, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of hank Sent: Tuesday, June 28, 2005 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt] how do they know if your calling your tax dude or something? what do they do monitor the calls or something? ----- Original Message ----- From: "John Goerzen" <jgoerzen@complete.org> To: <asterisk-users@lists.digium.com> Sent: Tuesday, June 28, 2005 6:46 AM Su...

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 great instructions (there weren't any) but a snappy response from the time I paid in paypa...