On 04/25/2012 05:29 PM, Eric Wieling wrote:
-----Original Message-----
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Wednesday, April 25, 2012 6:25 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hangup Cause and SIP Response Code
On 04/25/2012 04:45 PM, brya...@zktech.com wrote:
Kevin
I am using 1.8.x& 10.x
Then you have SIP_CAUSE available, although you'll have to enable it because it
is off by default due to performance concerns.
============================================
Does anyone know what kind of performance hit you take from SIP_CAUSE when you
are using few or no calls using chan_local?
The performance impact will be directly related to the number of
outbound SIP channels you create; no other channels will be involved. We
had a Digium OEM customer observe a 50% call load capability decrease
when they started using SIP_CAUSE, but that was on a pretty busy system,
and all the channels were SIP channels.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
--
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