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(closes issue #10569) 

Reported by: IgorG
Patches:
      sip_conf-80933-1.patch uploaded by IgorG (license 20)
Fix up sip.conf sample configuration.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@80962 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
file 2007-08-27 12:18:13 +00:00
parent 2397e363d1
commit 6b390a46f3
1 changed files with 2 additions and 8 deletions
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10
configs/sip.conf.sample
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@ -137,12 +137,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; 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 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;
; [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
; ...
;