On 04/25/2012 05:29 PM, Eric Wieling wrote:
-----Original Message----- From: asterisk-users-bounces@xxxxxxxxxxxxxxxx [mailto:asterisk-users-bounces@xxxxxxxxxxxxxxxx] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 25, 2012 6:25 PM To: asterisk-users@xxxxxxxxxxxxxxxx Subject: Re: Hangup Cause and SIP Response Code On 04/25/2012 04:45 PM, BryantZ@xxxxxxxxxx wrote:Kevin I am using 1.8.x& 10.xThen 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@xxxxxxxxxx | SIP: kpfleming@xxxxxxxxxx | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- 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
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