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author | mattf <mattf@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-01-20 23:19:49 +0000 |
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committer | mattf <mattf@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-01-20 23:19:49 +0000 |
commit | c9fafda4b1b1e4d3db0a9321ea1f61a1905dd4cc (patch) | |
tree | 8746caa21fe3b70e78b77ff27f4f58fae32fd01c /configs/sip.conf.sample | |
parent | 73a64f9573ac445e0bd36511e4f18c94b1b88270 (diff) |
Fix comments in sip.conf (#6134)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@8359 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs/sip.conf.sample')
-rw-r--r-- | configs/sip.conf.sample | 2 |
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 |