[asterisk-users] 1.6 and 1.8
Bruce B
bruceb444 at gmail.com
Thu Dec 29 11:18:55 CST 2011
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 <rswagoner at gmail.com> wrote:
> On Thu, Dec 29, 2011 at 12:05 AM, Bruce B <bruceb444 at gmail.com> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111229/c2c8d482/attachment.htm>
More information about the asterisk-users
mailing list