Hello Everyone,
We are getting some rather poor results (relative) with our Asterisk
setup. Not sure if we are using the sipp correctly etc.. but
nevertheless, is there any documentation that describes how we can get
the most our of our Asterisk box. For example when we hit the too
many file
Mitch Claborn wrote:
I get to go home on Saturday! The Digium phone deployment is simple
enough to manage remotely.
Glad to hear it. If the problem comes back on the hardphones, just post the
debug information to this thread and I'll take a look at it.
Regards,
Matthew Roth
InterMedia
I dont apply any secret recipe while installing asterisk, but maybe you can
share yours...
I am typing from my mobile phone...
Il giorno 23/mar/2013 14:34, Nick Khamis sym...@gmail.com ha scritto:
Hello Everyone,
We are getting some rather poor results (relative) with our Asterisk
setup. Not
Oh no secret. Some things I do is increase the ulimit size. I was
wondering if there was a way to increase allocated memory. I have been
reading about a -p option but when I start asterisk using asterisk -p
-10 it does not accept it but asterisk -p 10 works fine. Not sure
if that was the intended
On 03/22/2013 10:13 PM, Vladimir Mikhelson wrote:
Chris,
Thank you for sharing. It will help one day when 11 will become stable
enough to consider it for a production system.
Interestingly, jabber.conf has the same exact setting and the same exact
value and comment in 1.8.20.1:
priority = 1
Nick Khamis wrote:
Oh no secret. Some things I do is increase the ulimit size. I was
wondering if there was a way to increase allocated memory. I have been
reading about a -p option but when I start asterisk using asterisk -p
-10 it does not accept it but asterisk -p 10 works fine. Not sure
if
On Sat, Mar 23, 2013 at 12:06 PM, Joshua Colp jc...@digium.com wrote:
Nick Khamis wrote:
Oh no secret. Some things I do is increase the ulimit size. I was
wondering if there was a way to increase allocated memory. I have been
reading about a -p option but when I start asterisk using asterisk
Hello Gentlemen,
Thank you so much for your responses. We have been working on a
SIP/RTP Proxy+Asterisk in backed by MySQL for a few weeks. Everything
is working nicely I am pleased to say. And will be making some
donations for G729 licenses etc.. (it's the least we can do to support
the cause).
On Sat, Mar 23, 2013 at 3:21 PM, Nick Khamis sym...@gmail.com wrote:
Hello Gentlemen,
Thank you so much for your responses. We have been working on a
SIP/RTP Proxy+Asterisk in backed by MySQL for a few weeks. Everything
is working nicely I am pleased to say. And will be making some
donations
It isn't a donation, it is a licensing fee so you can legally transcode g729.
If Asterisk has to modify or generate an audio stream, then you need to
transcode.
Examples of this are early audio ringback, conferencing, playing back any audio
files which are not already in g729 format, I'm sure
Hello guys, no we do not do any recording of any kind. It was my
assumption that processing media in g729 requires some sort of
transcoding on the box?
N.
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On Sat, Mar 23, 2013 at 10:45 AM, Harley Peters
har...@thepetersclan.com wrote:
I had it set to 1 originally and it worked fine at first then suddenly
stopped.
It drove me crazy until I ran across this link:
http://iprouteth0.blogspot.com/2013/01/new-thoughts-troubleshooting-google.html
Set
On Sat, 23 Mar 2013 00:49:31 +, Jaap Winius wrote:
There is a file, called /etc/sysctl.d/bindv6only.conf, on my system that
sets /proc/sys/net/ipv6/bindv6only to 1 and it's been there since June
2010 when I installed Debian squeeze on my server machine (while squeeze
was still in its
On 3/23/2013 10:45 AM, Harley Peters wrote:
I'm running asterisk 1.8.10.1 and can confirm it works the same way.
I had it set to 1 originally and it worked fine at first then suddenly
stopped.
It drove me crazy until I ran across this link:
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