<div dir="ltr">I'm using it.<br><br>Can you please provide more information on the issue with this feature ? <br>Is there another way to know the response code of SIP ?<br><br>Thanks,<br><br>Ido<br><br><div class="gmail_quote">
On Thu, Aug 18, 2011 at 15:42, Matthew Nicholson <span dir="ltr"><<a href="mailto:mnicholson@digium.com">mnicholson@digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Greetings,<br>
<br>
Recently a performance regression in chan_sip was discovered in Asterisk<br>
1.8. The regression is caused by chan_sip setting<br>
MASTER_CHANNEL(HASH(SIP_CAUSE,<chan name>)) after each response received<br>
on a channel. That feature has been made optional in the latest 1.8 SVN<br>
code, but is currently still enabled by default. After some internal<br>
discussion, we decided to consider disabling this feature by default in<br>
future 1.8 versions. This would be an unexpected behavior change for<br>
anyone depending on that SIP_CAUSE update in their dialplan.<br>
Alternatively, with this feature enabled, anyone upgrading from Asterisk<br>
1.4 will see a 60% decrease in the amount of SIP traffic they can handle<br>
before encountering problems.<br>
<br>
Before disabling this feature, we wanted to get a feel for how many<br>
people are using it. If you use this feature, please respond to this<br>
email and let us know.<br>
--<br>
Matthew Nicholson<br>
Digium, Inc. | Software Developer<br>
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