similar to: Anyone having problems with LiveVoIP?

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?

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

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

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

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 --

> 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 hope this is an appropriate question for the list.. I am looking for a VOIP termination provider who can offer the following: -Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. So far I have tested 4 providers which I will not mention here. I have found two

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

Here is the call flow: [ivr-incoming] exten => s,1,LookupCIDName exten => s,2,DigitTimeout(2) exten => s,3,ResponseTimeout(10) exten => s,4,Wait(1) exten => s,5,Background(custom/ivr-incoming) exten => 1,1,Background(pls-wait-connect-call) exten => 1,2,Dial(${RINGPHONENUMBERS},20,r) exten => 1,3,Voicemail,u${VMBOX} exten => 1,4,Hangup Running * 1.0.5. The calling party

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?

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

Hi, I am looking for a provider that can provide DIDs in Atlanta, GA (national in future) and who can port an existing number. LiveVoip is a perfect fit but doesn't do LNP :-(. Currently I only need a handful of numbers ported, but will have more requirement in future. I'll need unlimited simultaneous incoming/outgoing calls (can pay for incoming as well). Any pointers are much

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

I agree with that fact the same questions get posted, but that problem is compounded by the fact the archives are not really searchable. If the were as lease some users would search. The archives need 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

Just a quick couple of questions for ya'll. 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones? I have had a terrible time getting a hold of anyone over there, and I need this functionality before I can migrate to * completely. 2) Are there currently any problems with inbound DID's? Everything is setup properly in *, but I am not able to receive inbound calls,

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

Nothing has changed in my Asterisk configuration and now outbound US is getting nothing, but 403's. Anyone else having the same problem? Inbound calls to my DID's are working fine. Thanks, SG -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070325/1f30a3d3/attachment.htm

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

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

All, I use various IAX providers to terminate my outbound calls. I set the caller-ID to one of several DIDs, based on the called number. There doesn't seem to be any rhyme or reason as to what the called user sees, however. Calls to most cell-phones show *exactly* the number I submit. Calls to land-lines sometimes show, other times not. Calls to other voip-providers most often show, but