diff options
| author | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-01-18 22:04:33 +0000 |
|---|---|---|
| committer | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-01-18 22:04:33 +0000 |
| commit | d6e19bdc91b0c4c6b5a069e11898741ec082b289 (patch) | |
| tree | d0cb360114e418a612eb2025d270801a1388cd7f /configs | |
| parent | cc1fcc753900c912d856f3f0498a4f7bfd8344a6 (diff) | |
Merge changes from team/group/sip-tcptls
This set of changes introduces TCP and TLS support for chan_sip. There are various
new options in configs/sip.conf.sample that are used to enable these features. Also,
there is a document, doc/siptls.txt that describes some things in more detail.
This code was implemented by Brett Bryant and James Golovich. It was reviewed
by Joshua Colp and myself. A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what exists there.)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99085 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
| -rw-r--r-- | configs/sip.conf.sample | 16 |
1 files changed, 14 insertions, 2 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index acca09c6c..62eee41bb 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -70,6 +70,16 @@ allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) ; bindport is the local UDP port that Asterisk will listen on bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) + +tcpenable=yes ; Enable server for incoming TCP connections (default is yes) +tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) + ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) + +;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no) +;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces) + ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061) +;tlscertfile=asterisk.pem ; Certificate file (*.pem only) to use for TLS connections + ; default is to look for "asterisk.pem" in current directory srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records @@ -320,7 +330,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------ ; Asterisk can register as a SIP user agent to a SIP proxy (provider) ; Format for the register statement is: -; register => user[:secret[:authuser]]@host[:port][/extension] +; register => [transport://]user[:secret[:authuser]]@host[:port][/extension] +; +; ; ; If no extension is given, the 's' extension is used. The extension needs to ; be defined in extensions.conf to be able to accept calls from this SIP proxy @@ -607,7 +619,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; User config options: Peer configuration: ; -------------------- ------------------- ; context context -; callingpres callingpres +; callingpres callingpres ; permit permit ; deny deny ; secret secret |