Asterisk 1.8 SIP_CAUSE performance regression

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Asterisk Users 1 Comment


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,)) 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.

One thought on - Asterisk 1.8 SIP_CAUSE performance regression

  • Hi,

    We’re using it here. As Ido asked, is there an alternative way of
    getting the SIP response in the event a Dial() fails?