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author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-01-24 18:15:20 +0000 |
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committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-01-24 18:15:20 +0000 |
commit | 295fbee42e2b6507348cce689e0eeb6509f15aa5 (patch) | |
tree | 7bc47931ae4a5f8a8764a0950515364295ae0420 /configs | |
parent | 353cec56ea1dea09a99bb960ebdd1f96a209190f (diff) |
Make it clear that caller ID in sip.conf is used only on incoming calls (inspired by bug #6183)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@8554 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
-rw-r--r-- | configs/sip.conf.sample | 1 |
1 files changed, 1 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 7a671b996..da63b4d5e 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -342,6 +342,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;type=friend ;context=from-sip ; Where to start in the dialplan when this phone calls ;callerid=John Doe <1234> ; Full caller ID, to override the phones config + ; on incoming calls to Asterisk ;host=192.168.0.23 ; we have a static but private IP address ; No registration allowed ;nat=no ; there is not NAT between phone and Asterisk |