On Mon, 31 Mar 2014, Shaun Ruffell wrote:
If you're looking to reduce the CPU overhead of processing meetme
conferences, this email from awhile ago may be of some help:
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/51750/focus=51777
Thanks for the clue. I can hit my target o
On Fri, Mar 21, 2014 at 11:26:22AM -0700, Steve Edwards wrote:
> On Fri, 21 Mar 2014, Steve Totaro wrote:
>
> >I found below here:
> > http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
> >
> >If you have too many conferences, one CPU may not be able to mix all the
> >audio and you will have a
2014-03-21 18:54, Steve Totaro skrev:
I found below here: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
If you have too many conferences, one CPU may not be able to mix all the
audio and you will have audio problems even if there are 7+ other CPUs
that are essentially idle while waiting
In article ,
Steve Edwards wrote:
> On Fri, 21 Mar 2014, Steve Edwards wrote:
>
> > Googling 'dahdi pseudo channel limit modprobe' showed the secret sauce.
>
> Oops. Guess I should complete the thread...
>
> You can set the DAHDI pseudo channel limit in /etc/modules.conf:
>
> options
sts.digium.com] On Behalf Of Steve Totaro
Sent: Friday, March 21, 2014 8:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Need more meetme users -- hitting some limit
Is there any good docu
On Fri, 21 Mar 2014, Steve Edwards wrote:
I'm trying to make the leap from 1.2 to 11.8.1
On Fri, 21 Mar 2014, Steve Totaro wrote:
Is there any good documentation on that process?
I haven't looked. I know they added a few of variables to the AGI
environment Asterisk passes to your AGI on
Is there any good documentation on that process?
On Fri, Mar 21, 2014 at 3:36 PM, John Novack
wrote:
>
> Steve Edwards wrote:
>
>> On Fri, 21 Mar 2014, Adrian Serafini wrote:
>>
>> Upgrade to 1.4? hehe, I thought you were the self proclaimed 1.2
>>> luddite? I'm a big fan of older releases wit
On Fri, 21 Mar 2014, Steve Edwards wrote:
Googling 'dahdi pseudo channel limit modprobe' showed the secret sauce.
Oops. Guess I should complete the thread...
You can set the DAHDI pseudo channel limit in /etc/modules.conf:
options dahdi max_pseudo_channels=x
or you can set it from t
On Fri, 21 Mar 2014, Paul Belanger wrote:
DAHDI has a pseudo channel limit of 512, somebody has already posted how
to change it with modprode.
Not in this thread, but big thanks for the clue. Googling 'dahdi pseudo
channel limit modprobe' showed the secret sauce.
I can get 1,000 simultaneou
On Fri, Mar 21, 2014 at 11:53 AM, Steve Edwards
wrote:
> I'm trying to determine the capacity of my host running Asterisk 11.8.1 on
> CentOS 6.5.
>
> The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet.
>
> The primary application will be bridging groups of users using meetme
Steve Edwards wrote:
On Fri, 21 Mar 2014, Adrian Serafini wrote:
Upgrade to 1.4? hehe, I thought you were the self proclaimed 1.2 luddite? I'm
a big fan of older releases with 1 year plus of uptime.
Yep, that's me :)
I'm trying to make the leap from 1.2 to 11.8.1
That is a HUGE leap
Wat
On Fri, 21 Mar 2014, Adrian Serafini wrote:
Upgrade to 1.4? hehe, I thought you were the self proclaimed 1.2
luddite? I'm a big fan of older releases with 1 year plus of uptime.
Yep, that's me :)
I'm trying to make the leap from 1.2 to 11.8.1
--
Thanks in advance,
--
On Fri, Mar 21, 2014 at 2:26 PM, Steve Edwards wrote:
> On Fri, 21 Mar 2014, Steve Totaro wrote:
>
> I found below here: http://www.voip-info.org/
>> wiki/view/Asterisk+cmd+MeetMe
>>
>> If you have too many conferences, one CPU may not be able to mix all the
>> audio and you will have audio prob
Coincidentally, 512 is my target. Any clues on how to get 200 more?
Upgrade to 1.4? hehe, I thought you were the self proclaimed 1.2
luddite? I'm a big fan of older releases with 1 year plus of uptime.
--
_
-- Bandwidt
On Fri, 21 Mar 2014, Steve Totaro wrote:
I found below here:
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
If you have too many conferences, one CPU may not be able to mix all the
audio and you will have audio problems even if there are 7+ other CPUs
that are essentially idle while
On (21/03/14 13:54), Steve Totaro put forth
the proposition:
I found below here: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
If you have too many conferences, one CPU may not be able to mix all the
audio and you will have audio problems even if there are 7+ other CPUs that
are esse
I found below here: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
If you have too many conferences, one CPU may not be able to mix all the
audio and you will have audio problems even if there are 7+ other CPUs that
are essentially idle while waiting for one CPU to mix everything. You
sho
On Fri, 21 Mar 2014, Steve Edwards wrote:
The call file on box1 originates a call to box2 and then plays a 2 hour
WAV file.
The call file on box1 originates a SIP call to box2 and then plays a 2
hour WAV file.
--
Thanks in advance,
---
I'm trying to determine the capacity of my host running Asterisk 11.8.1 on
CentOS 6.5.
The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet.
The primary application will be bridging groups of users using meetme().
I'm using 2 boxes -- 1 to initiate calls using call files (
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