similar to: VoipJet TOS (was Teliax and also LiveVoip)

So far my experience with TOS has been that most of them are pretty odd. No one wants the liability of a stock trade gone foul or a call to the doctor that gets disconnected. Essentially, those things in the TOS are just a CYA. They are un-enforced but should someone decide to attempt to sue based upon a financial loss, the ITSP is covered. So, yep. That is weird but not unexpected. Heaven

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:

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

<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

Hi, I am new to asterisk , i am getting the following error,& the /etc/zaptel.conf file entry is as follows defaultzone=us loadzone=us span=1,1,0,esf,b8zs,yellow bchan=1-23 dchan=24 Parsing '/etc/asterisk/zapata.conf': Found Jul 1 18:33:35 WARNING[16384]: chan_zap.c:664 zt_open: Unable to specify channel 1: No such device or address Jul 1 18:33:35 ERROR[16384]: chan_zap.c:5296

I'm trying to setup asterisk to accept call from Teliax, request the 10 digit number from user, then dial it thru the VoIPJet. If I'm not wrong I will be charged by both providers because both connection is active during conversation. So my question is can I set the things so that I pay only to VoIPJet? Specific configuration snippets will be greatly appeciated. Thank you.

I've heard good things about VoipJet here, so I was going to set up an account. Then I noticed their Terms of Service here: https://www.voipjet.com/tos.php Several things there are very concerning to me, and I'm interested in what other people here think of them. * The ToS specifically forbids use for any call relating to medical, financial, or government matters -- as well as any

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

We are looking for SIP trunks for our * pbx for our business. Being able to port our numbers is an absolute requirement. teliax can do it, but I am unsure of the others. Anyone have experiences (good, bad) with the above mentioned providers to share? Eg reliability, quality, etc. -Dan

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

Hello. I am having an issue where sometimes the cheapest provider for certain international destinations is not always reliable in completing calls. However, there is not problem once the call is made (i.e. no lag or echo or anything). The way I have it set up right now (for example) for Dar es Salaam, Tanzania is: exten => _925522XX.,1,Dial(IAX2/livevoip/011${EXTEN:1}) exten =>

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

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

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 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?

Whoa, talk about flying under the radar. I got a few DIDs *months* ago from Sixtel (or iax.cc). Initial respone was great, but then it seemed that the only tech-support person had fled the country. No responses, bad responses, poor call-quality. I had a few $s left in the balance and kind-a just forgot about them. Not worth 10 minutes of my time to get $10 of my money back, quite frankly.

TelIAX, one of the new VoIP-to-PSTN gateway providers, has their terms of service posted on their signup page: http://teliax.com/user_admin/signup/s1.php They look strangely familiar--it's exactly the same as http://www.vonage.com/features_terms_service.php with s/Vonage/Teliax/. (And it's cut off about halfway through). Anyone else notice this? Brad

> 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

Hi, I am Chandramouli from India. I have successfully implemented Intercom (Dialling within my office LAN) and Voicemail features using Asterisk. To implement this, I am using X-Lite Softphone. Now, I wish to make calls to US using VoIP Asterisk. I dont think I need any external hardware to implement pure VoIP solution. Here I am sending my configuration file values: Contents of