[asterisk-users] Hangup Cause and SIP Response Code

[Eric Wieling]

-----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.

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Does anyone know what kind of performance hit you take from SIP_CAUSE when you are using few or no calls using chan_local?