I'm using it. Can you please provide more information on the issue with this feature ? Is there another way to know the response code of SIP ?
Thanks, Ido On Thu, Aug 18, 2011 at 15:42, Matthew Nicholson <[email protected]>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. > -- > Matthew Nicholson > Digium, Inc. | Software Developer > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev >
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