[Asterisk-Users] Re: LiveVOIP

[Chris Coulthurst]

Did any Level 3 backbone go down about the same time?  Maybe some BGP4
routes got 'lifeboated'.

Chris Coulthurst
chris at shuksan.com
 

|-----Original Message-----
|From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
|bounces at lists.digium.com] On Behalf Of David Josephson
|Sent: Monday, May 02, 2005 8:18 PM
|To: asterisk-users at lists.digium.com
|Subject: [Asterisk-Users] Re: LiveVOIP
|
|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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