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authorMatthew Fredrickson <creslin@digium.com>2006-01-20 23:19:49 +0000
committerMatthew Fredrickson <creslin@digium.com>2006-01-20 23:19:49 +0000
commit4dc76fbcc15423310c0eed68d4c7af4c6166d69b (patch)
tree8746caa21fe3b70e78b77ff27f4f58fae32fd01c /configs/sip.conf.sample
parent52fe16259b23f5820b65486d82963e5b8b23851e (diff)
Fix comments in sip.conf (#6134)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'configs/sip.conf.sample')
-rw-r--r--configs/sip.conf.sample2
1 files changed, 1 insertions, 1 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 61ea9ba7a..7a671b996 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -90,7 +90,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;useragent=Asterisk PBX ; Allows you to change the user agent string
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
; Note that promiscredir when redirects are made to the
- ; local system will cause loops since SIP is incapable
+ ; local system will cause loops since Asterisk is incapable
; of performing a "hairpin" call.
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
; a valid phone number