Log are being filled with g729 transcoding error in 1.8.7x now :-( I don't dare to test 1.8.8x as it might have something else broken. Unfortunately I can no longer trust the release candidates. Thanks for the input.
On Thu, Dec 29, 2011 at 8:29 AM, Ryan Wagoner <[email protected]> wrote: > On Thu, Dec 29, 2011 at 12:05 AM, Bruce B <[email protected]> wrote: > >> I have been running 1.8.7 with a few fixes back ported from the 1.8.8 >>> release candidate for the last 2.5 months. The system processes around >>> 4,000 calls per day over PRIs for 250 Polycom phones. >>> >>> Previously I was running 1.6.1.18 with a bunch of back ports for fixes >>> and features. Overall it was stable but every few months I had an issue >>> where a channel would get hung. When this happened core show channels would >>> crash the console and I would eventually have to restart Asterisk. >>> >>> Ryan >>> >> >> What od you mean by, "been running 1.8.7 with a few fixes back ported >> from the 1.8.8 release candidate". So, this is a version 1.8.7 release that >> you are using or a 1.8.8 or is this a mix of both that you come up with? >> Can you please be specific with fixes? >> >> Thanks >> >> > It was a mix I came up with as I was hitting a few bugs in 1.8.7 and 1.8.8 > wasn't released. At this point I would just go for 1.8.8. The issue was > mainly 17541 which was filling my logs and basically made Asterisk unusable. > > https://issues.asterisk.org/jira/browse/ASTERISK-17541 > https://issues.asterisk.org/jira/browse/ASTERISK-18570 > https://issues.asterisk.org/jira/browse/ASTERISK-18101 > > I had tested 1.8.4 before and was hit by a bunch of dtmf issues that were > fixed in 1.8.5. When 1.8.7 came out it looked fairly stable so I switched > from 1.6.1.18. I was running the 1.6.1 branch as I needed TCP SIP support. > Right now I have been testing 1.8.8 which looks to be a good release. The > 1.8 series has come a long way in a few releases as far as fixing major > bugs. > > Ryan > > > -- > _____________________________________________________________________ > -- 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 >
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