[asterisk-commits] file: trunk r80962 - /trunk/configs/sip.conf.sample
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Aug 27 07:18:13 CDT 2007
Author: file
Date: Mon Aug 27 07:18:13 2007
New Revision: 80962
URL: http://svn.digium.com/view/asterisk?view=rev&rev=80962
Log:
(closes issue #10569)
Reported by: IgorG
Patches:
sip_conf-80933-1.patch uploaded by IgorG (license 20)
Fix up sip.conf sample configuration.
Modified:
trunk/configs/sip.conf.sample
Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=80962&r1=80961&r2=80962
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Mon Aug 27 07:18:13 2007
@@ -137,12 +137,6 @@
; for peers and users as well
;callevents=no ; generate manager events when sip ua
; performs events (e.g. hold)
-;limitonpeers=no ; Apply all call limits ("limit=") only to peers, never
- ; to users. This improves handling of call limits
- ; and device states in certain situations. The user part
- ; of a type=friend will still be affected by the call
- ; limit, but Asterisk will only use one object for
- ; counting the simultaneous calls.
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
; for any reason, always reject with '401 Unauthorized'
; instead of letting the requester know whether there was
@@ -669,9 +663,9 @@
;
; [2133](natted-phone,my-codecs)
; secret = peekaboo
-; [2134](natted-phone,ulaw-hone)
+; [2134](natted-phone,ulaw-phone)
; secret = not_very_secret
-; [2136](public-phone,ulaw-hone)
+; [2136](public-phone,ulaw-phone)
; secret = not_very_secret_either
; ...
;
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