On Sat, Jan 3, 2009 at 13:09, Trixter aka Bret McDanel <[email protected]> wrote: > On Sat, 2009-01-03 at 12:14 -0500, Andrew Joakimsen wrote: >> Can you look at ticket # 702556000194? >> >> This is very simple: >> > > apparently it isnt. > > >> Asterisk is down, I am simulating that with the command "stop now," >> Calls should then go to the failover SIP address, but they do not. >> >> I have been back and forth for weeks with your support and they do not >> figure it out. I am not even sure they understand what I am saying. >> >> > > is this related to the below request for a non-profit doing a telethon? > If it is I am confused by it. > > If it isnt, I am unsure what ticket system you refer to. Additionally I > am unsure what your setup is since you havent even provided more > information. Odds are the equipment that is supposed to do the failover > isnt even asterisk. Further I do not think that its a business list > question (unless you are asking for a consultant to fix your problem > with failover). > > Or was this supposed to be a private email to someone at supertec that > you posted to a public mailing list? > > >> >> On Tue, Dec 9, 2008 at 15:34, Suzanne Bowen <[email protected]> wrote: >> > There is a Northwest Florida organization http://www.linkingarms.org who >> > wants to have a telethon using open source telephony technologies. If >> > anyone >> > reading this is interested in talking with the executive director Kenny >> > about this, please email me OFF the list so we won't bother the list with >> > further details. >> > >
You are right, the message was sent to the wrong person. But now that this is out in the open, let me explain: DIDx has an "alternate ring-to" feature. So if our Asterisk server is down, the calls can roll-over elsewhere. This feature is not working. The calls do roll-over, but there is no audio (even using DIDx's own "carrierx.us" proxy) and drop a few seconds later. Testing from a landline, the caller hears a few seconds of silence and then a reorder tone. If we set the "alternate ring-to" proxy as the primary ring-to, it does work. Clearly the issue is on DIDx's end. I conducted my testing by issuing the "stop now" command at the CLI and called the DID number from an AT&T landline phone. I was promised last week this would be looked into, fixed, etc. But no response thusfar. I am going to stop using DIDx as it has been one big headache. On top of this not working, the CDRs on their website are incorrect -- either they have inept programmers or they do absolutely no QA testing on their code -- but they'll gladly sell you a copy for upwards of $20,000. _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
