[asterisk-dev] Suspected deadlocks in Asterisk 1.8 under heavy load

[Kevin P. Fleming]

On 08/16/2011 09:27 PM, Alistair Cunningham wrote:
> On 17/08/11 12:23, Kevin P. Fleming wrote:
>> Matt Nicholson committed a change to the 1.8, 10 and trunk branches
>> today to solve a significant performance issue caused by the change to
>> chan_sip to return the SIP hangup cause to the 'master' channel. His
>> change made that behavior optional, even though it was already released
>> in 1.8, because of the performance impact it has. We had another
>> customer report a similar set of symptoms.
>>
>> If possible, it would be most helpful if you could try that patch on one
>> of your affected systems before you downgrade it. I can understand if
>> your customer is not willing to let you try that, though :-)
>
> Kevin,
>
> Thank you for this. When you say "optional", are any configuration
> settings needed to disable it?

Yes, in Asterisk 1.8 you'll need to set 'storesipcause' to 'off', since 
the default is 'on' to preserve the existing behavior. In Asterisk 10 
and later, the default will be 'off'.

If we have real-world testing that shows that changing the default to 
'off' will resolve issues such as yours, we'll consider even changing 
the default to 'off' for the Asterisk 1.8.6 release. It'd be an unusual 
step to take, but unless there is sufficient community demand for this 
feature to be enabled by default, the performance problems it causes are 
not acceptable for users who don't care about the feature.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org