[Asterisk-Users] Re: LiveVOIP

[David Josephson]

Luki writes about choppy audio with LiveVOIP. We have an almost 
identical situation except that we were switched from the San Diego 
gateway to the Van Nuys gateway. Some improvement but still not usable 
for real customers. I have an open trouble ticket with them and no 
progress. Doesn't matter whether it's MOH, IVR audio or calls; incoming 
audio and DTMF dialing is fine, outgoing audio to the PSTN is choppy, at 
best one dropout every 10 seconds, usually one short dropout every one 
to three seconds. The comments from their tech support and CTO were that 
they were aware of the problem and it was "a capacity issue" that they 
were working on. There is a separate problem in that ringback tone (or 
any other audio sent without answer supervision being active, 
apparently) is not played to the PSTN side. This is not unique to 
LiveVOIP and has been discussed (with its workarounds) before. I don't 
mind their brusque attitude or the lack of user-level support, but we 
won't be able to use their service if they can't fix the dropouts. There 
is a lot of clatter here on the list about them not being a "real 
provider" but a lot of this is sour grapes from people reselling more 
expensive service. We'll see ... they don't have to be 100% facilities 
based to provide good service, but they do have to fix this issue.
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