From 63de8343958b91c8836c5e6ddf1c0106b40e9fe6 Mon Sep 17 00:00:00 2001 From: Joshua Colp Date: Thu, 2 Apr 2009 17:20:52 +0000 Subject: 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 --- configs/sip.conf.sample | 2 ++ 1 file changed, 2 insertions(+) (limited to 'configs') 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 -- cgit v1.2.3