diff options
| author | Alexandr Anikin <may@telecom-service.ru> | 2012-07-04 21:42:05 +0000 |
|---|---|---|
| committer | Alexandr Anikin <may@telecom-service.ru> | 2012-07-04 21:42:05 +0000 |
| commit | fa10f3f8a8b727887ffd7857cd7eef222cb24efd (patch) | |
| tree | b84e871dbf3e36a4a35c0d550198c7b051936539 /configs | |
| parent | 50765000e619349e6492238197ffa854128b9329 (diff) | |
Added direct media support to ooh323 channel driver
options are documented in config sample
sample config rename to proper name - ooh323.conf
To change media address ooh323 send empty TCS if there was
completed TCS exchange or send facility forwardedelements
with new fast start proposal if not.
Then close transmit logical channels and renew TCS exchange.
If new fast start proposal is received then ooh323 stack call back
channel driver routine to change rtp address in the rtp instance.
If empty TCS is received then close transmit logical channels and
renew TCS exchange
Review: https://reviewboard.asterisk.org/r/1607/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'configs')
| -rw-r--r-- | configs/ooh323.conf.sample (renamed from configs/chan_ooh323.conf.sample) | 34 |
1 files changed, 24 insertions, 10 deletions
diff --git a/configs/chan_ooh323.conf.sample b/configs/ooh323.conf.sample index a39a47afc..d8ccdcf00 100644 --- a/configs/chan_ooh323.conf.sample +++ b/configs/ooh323.conf.sample @@ -27,13 +27,13 @@ ; OOH323/exten/peer OR OOH323/exten@ip ; ; Domain name resolution is not yet supported. -; +; ; When a H.323 user calls into asterisk, his H323ID is matched with the profile ; name and context is determined to route the call ; -; The channel driver will register all global aliases and aliases defined in +; The channel driver will register all global aliases and aliases defined in ; peer profiles with the gatekeeper, if one exists. So, that when someone -; outside our pbx (non-user) calls an extension, gatekeeper will route that +; outside our pbx (non-user) calls an extension, gatekeeper will route that ; call to our asterisk box, from where it will be routed as per dial plan. @@ -47,9 +47,9 @@ ;The dotted IP address asterisk should listen on for incoming H323 ;connections ;Default - tries to find out local ip address on it's own -bindaddr=0.0.0.0 +bindaddr=0.0.0.0 -;This parameter indicates whether channel driver should register with +;This parameter indicates whether channel driver should register with ;gatekeeper as a gateway or an endpoint. ;Default - no ;gateway=no @@ -65,7 +65,7 @@ bindaddr=0.0.0.0 ;H323-ID to be used for asterisk server ;Default - Asterisk PBX -h323id=ObjSysAsterisk +h323id=ObjSysAsterisk e164=100 ;CallerID to use for calls @@ -88,7 +88,7 @@ gatekeeper = DISABLE ;Sets default context all clients will be placed in. ;Default - default -context=public +context=default ;Sets rtptimeout for all clients, unless overridden ;Default - 60 seconds @@ -128,7 +128,7 @@ dtmfmode=rfc2833 ; CNG tone or an incoming T.38 RequestMode packet ; ; yes - enable both detection (CNG & T.38) -; no - disable both +; no - disable both ; cng - enable CNG detection (default) ; t38 - enable T.38 request detection ; @@ -137,7 +137,7 @@ faxdetect = cng ; User/peer/friend definitions: ; User config options Peer config options ; ------------------ ------------------- -; context +; context ; disallow disallow ; allow allow ; accountcode accountcode @@ -153,6 +153,20 @@ faxdetect = cng ; +; +; direct rtp between two remote peers, disabled by default +; can be specified globally or per peer/user section +; +directmedia=no +; +; early direct rtp (try to establish direct rtp before asnwer) +; disabled by default, auto enabled by directmedia is enabled +; can be disabled escpecially if directmedia is enabled. +; can be specified globally or per peer/user section +; +; +directrtpsetup=no + ;Define users here ;Section header is extension [myuser1] @@ -160,7 +174,7 @@ type=user context=context1 disallow=all allow=gsm -allow=ulaw +allow=ulaw |