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authoroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2006-10-17 17:39:18 +0000
committeroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2006-10-17 17:39:18 +0000
commit7268d40ab9da9f123c885756ab7db48817efe657 (patch)
tree826db3030db664bca56a04fc194d12b03b9017a9 /configs
parent5a4a0f68af54b4ff6e84a5049b793764f3b9df95 (diff)
Adding information about Marks direct-RTP hack to the docs...
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@45329 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
-rw-r--r--configs/sip.conf.sample5
1 files changed, 5 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index b101b9058..ef4fa2cda 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -256,6 +256,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; behind a NAT, or for some other reason wants Asterisk to
; stay in the audio path, you may want to turn this off.
+ ; In Asterisk 1.4 this setting also affect direct RTP
+ ; at call setup (a new feature in 1.4 - setting up the
+ ; call directly between the endpoints instead of sending
+ ; a re-INVITE).
+
;canreinvite=nonat ; An additional option is to allow media path redirection
; (reinvite) but only when the peer where the media is being
; sent is known to not be behind a NAT (as the RTP core can