[asterisk-users] Asterisk 1.8 SIP_CAUSE performance regression

[Kingsley Tart]

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.