path: root/README
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authormarkster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>2005-04-17 17:39:14 +0000
committermarkster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>2005-04-17 17:39:14 +0000
commita290adc7c5a2aaefe0958701d7f83fdbb510d831 (patch)
treee6b89f4bff5ff60d43c7c0249c2e70ca873df489 /README
parent220ac9b141f5910956a51a4abc441428f6c62ae4 (diff)
Note that big time changes can cause issues (bug #4020)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@5482 f38db490-d61c-443f-a65b-d21fe96a405b
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@@ -162,6 +162,23 @@ parameters. For example, the line "channel => 25-47" creates objects for
the channels 25 through 47 of the card, obtaining the settings
from the variables specified above.
+ Those using SIP phones should be aware the Asterisk is sensitive to large
+jumps in time. Those who live in areas that are on Daylight Savings Time (or
+equivalent) should set their system and hardware clocks to use UTC in order
+to avoid any possible jumps in system time. There should be no noticeable
+effects to the user, as you should still set your system to use the local
+offset from UTC.
+ Even for those who don't live in DST zones, this issue may manifest itself
+if the administrator makes large manual time adjustments. Thus, it is good
+practice to keep the time on your Asterisk server synced to a reliable
+source, such as an NTP server.
+ Also note that this issue is separate from the clocking of TDM channels, and
+is known to at least affect SIP registrations.
See the doc directory for more documentation.