Hi, We're using it here. As Ido asked, is there an alternative way of getting the SIP response in the event a Dial() fails?
Cheers, Kingsley. On Thu, 2011-08-18 at 07:42 -0500, Matthew Nicholson wrote: > Greetings, > > Recently a performance regression in chan_sip was discovered in Asterisk > 1.8. The regression is caused by chan_sip setting > MASTER_CHANNEL(HASH(SIP_CAUSE,<chan name>)) after each response received > on a channel. That feature has been made optional in the latest 1.8 SVN > code, but is currently still enabled by default. After some internal > discussion, we decided to consider disabling this feature by default in > future 1.8 versions. This would be an unexpected behavior change for > anyone depending on that SIP_CAUSE update in their dialplan. > Alternatively, with this feature enabled, anyone upgrading from Asterisk > 1.4 will see a 60% decrease in the amount of SIP traffic they can handle > before encountering problems. > > Before disabling this feature, we wanted to get a feel for how many > people are using it. If you use this feature, please respond to this > email and let us know. -- _____________________________________________________________________ -- 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
