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authorfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2009-04-02 17:20:52 +0000
committerfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2009-04-02 17:20:52 +0000
commit0eb1480fe02b28de9d0d67bbd8779d7296639cc1 (patch)
tree8a8042738e1c444e5988a648b795c4d2b02febd1 /configs/sip.conf.sample
parent889f2ce31ec2f6cda98ecbc9681b883b7384fa2c (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: http://svn.digium.com/svn/asterisk/trunk@186078 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs/sip.conf.sample')
-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