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.
David, The fact of the matter is that LiveVoIP has no customer service. They don't care about small users or asterisk users. Other providers have higher prices but offer real customer service. Go look at teliax they care about customers service. On Mon, 2005-05-02 at 22:17, David Josephson wrote:> 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. > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Did any Level 3 backbone go down about the same time? Maybe some BGP4 routes got 'lifeboated'. Chris Coulthurst chris@shuksan.com |-----Original Message----- |From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- |bounces@lists.digium.com] On Behalf Of David Josephson |Sent: Monday, May 02, 2005 8:18 PM |To: asterisk-users@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. |_______________________________________________ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users