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authorJoshua Colp <jcolp@digium.com>2009-04-02 17:20:52 +0000
committerJoshua Colp <jcolp@digium.com>2009-04-02 17:20:52 +0000
commit63de8343958b91c8836c5e6ddf1c0106b40e9fe6 (patch)
tree8a8042738e1c444e5988a648b795c4d2b02febd1 /configs
parent08971ce2056f4e035b4b37324c7f184370cd0ec6 (diff)
Merge in the RTP engine API.
This API provides a generic way for multiple RTP stacks to be integrated into Asterisk. Right now there is only one present, res_rtp_asterisk, which is the existing Asterisk RTP stack. Functionality wise this commit performs the same as previously. API documentation can be viewed in the rtp_engine.h header file. Review: http://reviewboard.digium.com/r/209/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'configs')
-rw-r--r--configs/sip.conf.sample2
1 files changed, 2 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 37fcb7405..3785618e3 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -292,6 +292,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
; register their phones.
+;engine=asterisk ; RTP engine to use when communicating with the device
+
;
; If regcontext is specified, Asterisk will dynamically create and destroy a
; NoOp priority 1 extension for a given peer who registers or unregisters with