Thanks for feedback. Yeah, tell me about it. Your description is very accurate of the situation. I can't believe it's in the repo without any tests done; even the simplest reload. I don't mean to be a whiner but honestly the repo is a joke with such an obvious flaw for so long now........
On Sun, Jul 31, 2011 at 4:03 AM, Vahan Yerkanian <va...@arminco.com> wrote: > On 7/30/11 7:39 AM, Bruce B wrote: > >> I think this should be a quick fix since it's rendering the latest >> stable version useless and making the impression that it was released >> just to break things and force people onto 1.8x. Just a thought...no >> blame game. But really something like this should be tackled quickly. No >> point to break things so badly on the last stable version. >> >> Regards, >> >> > Confirming this issue on 10+ 1.6.2.19 Asterisk servers. This problem makes > it difficult to do edits on sip.conf on production systems, as there is ~25% > chance that you'll crash the server and cut the established calls. The > problem does not exist in 1.6.2.18... > > I think this problem should be fixed or the 1.6.2.19 should be removed from > the digium repo. > > Regards, > Vahan > > > -- > ______________________________**______________________________**_________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users> >
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