[asterisk-dev] Asterisk 1.8 SIP_CAUSE performance regression

[ik]

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 <mnicholson at digium.com>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
>
>
>
>
> --
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