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author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-04-02 17:20:52 +0000 |
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committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-04-02 17:20:52 +0000 |
commit | 0eb1480fe02b28de9d0d67bbd8779d7296639cc1 (patch) | |
tree | 8a8042738e1c444e5988a648b795c4d2b02febd1 /configs/sip.conf.sample | |
parent | 889f2ce31ec2f6cda98ecbc9681b883b7384fa2c (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.sample | 2 |
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 |