diff options
author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-02-01 13:23:59 +0000 |
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committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-02-01 13:23:59 +0000 |
commit | 33d610bf382cec30b6d10292d85cf0dcfa2cb3a4 (patch) | |
tree | 718918ccaa260c43f69ad44d50e690e062f6b8a4 /configs | |
parent | 83da25ed32a34ea78ad18208a99bcee77e7e4691 (diff) |
- Clarify default setting of canreinvite (thanks royk)
- Add some extra headers and reference to other doc/ files for realtime
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9034 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
-rw-r--r-- | configs/sip.conf.sample | 20 |
1 files changed, 18 insertions, 2 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index d334bfb65..72bfcb5a8 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -109,6 +109,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Useful to limit subscriptions to local extensions ; Settable per peer/user also ;notifyringing = yes ; Notify subscriptions on RINGING state +;callevents=no ; generate manager events when sip ua performs events (e.g. hold) ; ; If regcontext is specified, Asterisk will dynamically create and destroy a @@ -119,6 +120,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ;regcontext=sipregistrations ; +;----------------------------------------- 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] @@ -152,7 +154,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; 0 = continue forever, hammering the other server until it ; accepts the registration ; Default is 0 tries, continue forever -;callevents=no ; generate manager events when sip ua performs events (e.g. hold) ;----------------------------------------- NAT SUPPORT ------------------------ ; The externip, externhost and localnet settings are used if you use Asterisk @@ -191,6 +192,21 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; route = Assume NAT, don't send rport ; (work around more UNIDEN bugs) +;canreinvite=yes ; Asterisk by default tries to redirect the + ; RTP media stream (audio) to go directly from + ; the caller to the callee. Some devices do not + ; support this (especially if one of them is + ; behind a NAT). + ; The default setting is YES. If you have all clients + ; behind a NAT, or for some other reason wants + ; Asterisk to stay in the audio path, + ; you may want to turn this off + +;----------------------------------------- REALTIME SUPPORT ------------------------ +; For additional information on ARA, the Asterisk Realtime Architecture, +; please read README.realtime and README.extconfig in the /doc directory of the +; source code. +; ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis? (yes|no) @@ -199,7 +215,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; If set to yes, when a SIP UA registers successfully, the ip address, ; the origination port, the registration period, and the username of ; the UA will be set to database via realtime. If not present, defaults to 'yes'. - ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule ; as if it had just registered? (yes|no|<seconds>) ; If set to yes, when the registration expires, the friend will vanish from @@ -220,6 +235,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; memory (due to caching or other reasons), the information will not be ; removed from realtime storage +;----------------------------------------- SIP DOMAIN SUPPORT ------------------------ ; Incoming INVITE and REFER messages can be matched against a list of 'allowed' ; domains, each of which can direct the call to a specific context if desired. ; By default, all domains are accepted and sent to the default context or the |