On 03/23/2014 11:26 AM, Monty Taylor wrote:
> Now - off to fix the nodes that are reporting in but are showing errors...
Patch for the pbx.openstack.org error:
https://review.openstack.org/82765
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st bet.
>
Good points ... I don't think any of the current gantt-core are
interested in reviewing right now. I'm happy to open up gantt-core to
Don and whoever else would like to join him, but I would want to move it
first.
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On 02/19/2014 05:58 AM, Thierry Carrez wrote:
> Russell Bryant wrote:
>> The gantt repository was originally put under openstack/. I would like
>> to move gantt to stackforge to better reflect the current status of the
>> project.
>>
>> We don't actually
de of Nova before
we can regenerate the gantt repo and try for the forklift again. In the
meantime, there are some that are still interested in experimenting with
the current gantt repo.
How do we go about doing this move? Should I file a bug to track it?
What can I do to help?
Thanks,
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hould be able to restore their abandoned change. In
> this case I've gone ahead as a Gerrit administrator and restored
> them for you.
>
Is it possible to grant this permission to foo-core?
I think it'd be helpful to allow all of n
one of the issues seems to be thrashing with our pool of
nodes. One thought for a seemingly simple change was to just put a
hard cap on how many changes on the gate queue get nodes allocated to
them at any given time so that the impact of a gate reset is lower.
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-BEGIN
be used across all of IBM, or should be the
name be more specific to what your team is testing?
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;s the test environment and
what patches are you testing against? Is this for testing changes
against your Neutron plug-in?
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comment includes a private address. Is this going to get updated so
that all of the status and results are public?
Build Started http://10.132.98.47/job/VCDriverTempestTests/154/
Thanks,
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can find the sweet spot.
>
> Shall we schedule another interactive session for this Friday at the
> usual 1700 UTC?
Fine with me.
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ormance hit.
The denoise audio filter works quite well in my experience. As the
description suggests, it does consume CPU, but it may be worth a shot.
Note that it requires a module that we don't have (func_speex, not
codec_speex, that's a typo ... I should go fix that).
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then switching to
> SIP if we had time / wanted a 2nd snapshot.
It shouldn't make a difference load-wise since it's a SIP call to the
server either way. The most reliable test would be using a Rackspace
instance in the same region as the server we're loading (Chicago
On 08/15/2013 07:02 PM, Paul Belanger wrote:
> On Tue, Aug 13, 2013 at 5:54 PM, Russell Bryant wrote:
>> On 08/13/2013 05:35 PM, James E. Blair wrote:
>>> Hi,
>>>
>>> On Friday, August 16th at 1700 UTC, we'd like to get a bunch of people
>>> calli
less SIP handling, and
> have reported them.
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he server to create load. We can pretty easily have calls come in that
randomly inject a sound prompt here and there.
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elp here. I want to learn more puppet anyway.
What kind of timeline are we shooting for here? If we want something
running sooner, could we start with a hacky approach for the Asterisk
configs (get puppet to place a set of files in /etc/asterisk as-is), and
the
On 07/01/2013 12:29 PM, Paul Belanger wrote:
> On Mon, Jul 1, 2013 at 11:31 AM, Russell Bryant wrote:
>> On 06/30/2013 01:03 PM, Paul Belanger wrote:
>>> Personally, I would stay with Asterisk 1.8, but that is just my
>>> opinion. WebRTC support in Asterisk is st
reference from them for a
>> couple of good VOIP providers -- when we get closer to having something
>> ready, I'll look into setting up an account.
>>
>
> I've used both http://voip.ms and http://flowroute.com in the past and
> have had no real issues.
Same.
> [1]
> https://github.com/kickstandproject/puppet-modules/tree/master/modules/asterisk
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