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Base memory usage

Asterisk gurus….

I just installed asterisk 1.8.1.1 along with FreePBX on a fairly small
VPS (512mb standard, 512mb “burst”). I note that the asterisk process
is using about 209mb of memory just doing nothing (not configured to do
anything yet)

In contrast to this, my 1.6.1.2 installation from a little over a year
ago uses only 40mb and it’s fully configured and running with about 4
months of uptime (2 trunks, 4 channels, 3 DIDs, and 4 extensions.)

Any ideas on how I can get the memory consumption down on my new
installation, or is it time to “downgrade” to the older version?

Thanks,
Larry

Users of CEL Please comment on Bug

If you are using CEL in asterisk 1.8 can you please look at the issue
tracker and comment.
On how this might effect you.

https://issues.asterisk.org/view.php?id=18559

Thanks
Bryant

Find media and sip endpoints IP address durring”h” extension

_____

From: href=”mailto:asterisk-users-bounces@lists.digium.com”>asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Thursday, December 30, 2010 2:32 PM
To: Asterisk Users Mailing List – Non-Commercial Discussion
Subject: [asterisk-users] Find media and sip endpoints IP address durring”h”
extension

How can I get the media and sip endpoints IP address durring “h” extension?

I need to write these to my CEL logs.

Any ideas?

Thanks
Bryant

Possibly from the SIP headers. Probably have to capture this information
before the hangup occurs (DeadAGI might be able to extract this).

Find media and sip endpoints IP address durring “h” extension

How can I get the media and sip endpoints IP address durring “h”
extension?

I need to write these to my CEL logs.

Any ideas?

Thanks
Bryant

The Year in VoIP

On this weeks VUC call we will collectively be our own guests. That is,
we’d like to know what was the big issue that impacted YOU in 2010? All
opinions welcome.

Here are a few things to get you thinking in advance:

– Apple’s Antenna-gate
– Asterisk 1.8 Launches
– Amazon EC2 as a DOS platform
– Cisco launched UMI video conference device
– More HDVoice capable phones
– Skype Outage
– VoIP on mobile devices
– or perhaps something more personal…..

Come one, come all. Bring your story.

Connect details at http://vuc.me

Michael Graves

Force different codecs on call base

Hello,

what i want to do is to find a way how i can solve the following problem.

we want to offer our customers in the country side also isdn over voip
but we have to use internet connections from another company for this.
This company offers a QoS on this connections but only with 192kbit
bandwith and with the ATM headers a normal g711a call has exactly 103,5
kbit/s so we can only use 1 channel but for isdn we need 2 :(

my idea was if i can find a way that the first call of a peer has g711a
codec (like normally) and if a second call comes in, or has to be placed
for this peer i only offer g726 (40kbit) so i dont have a bandwith issue.

is there a possible way of doing this or would it be easier to use two
peers, one with g711a and one with g726 and just let both only use one
channel?

thanks for your help!

best regards
stefan