[asterisk-users] [asterisk-dev] Asterisk 1.8 SIP_CAUSE performance regression

ik idokan at gmail.com
Thu Aug 18 07:49:30 CDT 2011


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
>
>
>
>
> --
> _____________________________________________________________________
> -- 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110818/4373e864/attachment.htm>


More information about the asterisk-users mailing list