<span style="font-family: Arial, Helvetica, sans-serif; font-size: 10pt"><span style="font-family: tahoma,arial,sans-serif; font-size: 10pt;"> <hr width="100%" size="2" align="center" />
<b>From</b>: "ik" <idokan@gmail.com><br />
<b>Sent</b>: Thursday, August 18, 2011 9:47 AM<br />
<b>To</b>: "Asterisk Developers Mailing List" <asterisk-dev@lists.digium.com><br />
<b>Subject</b>: Re: [asterisk-dev] Asterisk 1.8 SIP_CAUSE performance regression</span><br />
<br />
<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="border-left: #ccc 1px solid; margin: 0px 0px 0px 0.8ex; 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="border-left: rgb(204,204,204) 1px solid; margin: 0pt 0pt 0pt 0.8ex; 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 />
Does it give the same data as the hangupcause when you use the h context or is it different?<br />
<br />
Bryant</div>
</div>
</blockquote></div>
</div></span>