<div dir="ltr">On Thu, Aug 18, 2011 at 16:31, Matthew Nicholson <span dir="ltr"><<a href="mailto:mnicholson@digium.com">mnicholson@digium.com</a>></span> wrote:<br><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
When chan_sip sets MASTER_CHANNEL(HASH(SIP_CAUSE,<chan name>)) the<br>
MASTER_CHANNEL function scans the channel list which can be a slow<br>
operation. Under moderate traffic loads this causes delays in processing<br>
sip messages which can cause retransmission timers to expire.<br>
<br>
There is no other way to know the response code of a SIP message, but<br>
depending on what you are using the response codes for, you may be able<br>
to accomplish the same thing a different way. What are you using the<br>
codes for?<br></blockquote><div><br>I write application using Asterisk, and I use it to better understand the hangup cause, or error messages from SIP trunks. <br><br> <br>Ido<br></div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<br>
Also be aware that this feature is not going away, it simply will be<br>
disabled by default.<br>
<div><div></div><div class="h5"><br>
On Thu, 2011-08-18 at 15:49 +0300, ik wrote:<br>
> I'm using it.<br>
><br>
> Can you please provide more information on the issue with this<br>
> feature ?<br>
> Is there another way to know the response code of SIP ?<br>
><br>
> Thanks,<br>
><br>
> Ido<br>
><br>
> On Thu, Aug 18, 2011 at 15:42, Matthew Nicholson<br>
> <<a href="mailto:mnicholson@digium.com">mnicholson@digium.com</a>> wrote:<br>
> Greetings,<br>
><br>
> Recently a performance regression in chan_sip was discovered<br>
> in Asterisk<br>
> 1.8. The regression is caused by chan_sip setting<br>
> MASTER_CHANNEL(HASH(SIP_CAUSE,<chan name>)) after each<br>
> response received<br>
> on a channel. That feature has been made optional in the<br>
> latest 1.8 SVN<br>
> code, but is currently still enabled by default. After some<br>
> internal<br>
> discussion, we decided to consider disabling this feature by<br>
> default in<br>
> future 1.8 versions. This would be an unexpected behavior<br>
> change for<br>
> anyone depending on that SIP_CAUSE update in their dialplan.<br>
> Alternatively, with this feature enabled, anyone upgrading<br>
> from Asterisk<br>
> 1.4 will see a 60% decrease in the amount of SIP traffic they<br>
> can handle<br>
> before encountering problems.<br>
><br>
> Before disabling this feature, we wanted to get a feel for how<br>
> many<br>
> people are using it. If you use this feature, please respond<br>
> to this<br>
> email and let us know.<br>
> --<br>
> Matthew Nicholson<br>
> Digium, Inc. | Software Developer<br>
><br>
><br>
><br>
><br>
> --<br>
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</div></div>--<br>
<div><div></div><div class="h5">Matthew Nicholson<br>
Digium, Inc. | Software Developer<br>
<br>
<br>
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