-----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? -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
