[asterisk-users] Hangup Cause and SIP Response Code

Kevin P. Fleming kpfleming at digium.com
Fri Apr 27 12:34:32 CDT 2012

On 04/25/2012 05:29 PM, Eric Wieling wrote:
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin P. Fleming
> Sent: Wednesday, April 25, 2012 6:25 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Hangup Cause and SIP Response Code
> On 04/25/2012 04:45 PM, BryantZ at 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: kfleming at digium.com | SIP: kpfleming at 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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