Rename 'canreinvite' option to 'directmedia', with backwards compatibility.

It is clear from multiple mailing list, forum, wiki and other sorts of posts
that users don't really understand the effects that the 'canreinvite' config
option actually has, and that in some cases they think that setting it to 'no'
will actually cause various other features (T.38, MOH, etc.) to not work properly,
when in fact this is not the case. This patch changes the proper name of the
option to what it should have been from the beginning ('directmedia'), but
preserves backwards compatibility for existing configurations.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210190 65c4cc65-6c06-0410-ace0-fbb531ad65f3

4 files changed, 22 insertions, 22 deletions

diff --git a/configs/mgcp.conf.sample b/configs/mgcp.conf.sample
index 116b66cd0..fde0a4fc6 100644
--- a/configs/mgcp.conf.sample
+++ b/configs/mgcp.conf.sample
@@ -41,7 +41,7 @@
 ;[dlinkgw]
 ;host = 192.168.0.64
 ;context = default
-;canreinvite = no
+;directmedia = no
 ;line => aaln/2
 ;line => aaln/1
 
@@ -96,7 +96,7 @@
 ;callwaiting	= no
 ;callreturn	= yes
 ;cancallforward = yes
-;canreinvite	= no
+;directmedia	= no
 ;transfer	= no
 ;dtmfmode	= inband
 ;line => aaln/1			; now lets save this config to line1 aka aaln/1
@@ -104,7 +104,7 @@
 ;callwaiting	= no
 ;callreturn	= yes
 ;cancallforward = yes
-;canreinvite	= no
+;directmedia	= no
 ;transfer	= no
 ;dtmfmode	= inband
 ;line => aaln/2			; now lets save this config to line2 aka aaln/2
diff --git a/configs/res_ldap.conf.sample b/configs/res_ldap.conf.sample
index 0a442298d..b02045f15 100644
--- a/configs/res_ldap.conf.sample
+++ b/configs/res_ldap.conf.sample
@@ -60,7 +60,7 @@ name = cn
 amaflags = AstAccountAMAFlags
 callgroup = AstAccountCallGroup
 callerid = AstAccountCallerID
-canreinvite = AstAccountCanReinvite
+directmedia = AstAccountDirectMedia
 context = AstAccountContext
 dtmfmode = AstAccountDTMFMode
 fromuser = AstAccountFromUser
@@ -131,7 +131,7 @@ additionalFilter=(objectClass=*)
 amaflags = AstAccountAMAFlags
 callgroup = AstAccountCallGroup
 callerid = AstAccountCallerID
-canreinvite = AstAccountCanReinvite
+directmedia = AstAccountDirectMedia
 context = AstAccountContext
 dtmfmode = AstAccountDTMFMode
 fromuser = AstAccountFromUser
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index fef5ef8f4..ba9b0c619 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -662,17 +662,17 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;        nat = comedia           ; Use rport if the remote side says to use it and perform symmetric RTP.
 
 ;----------------------------------- MEDIA HANDLING --------------------------------
-; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
+; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
 ; no reason for Asterisk to stay in the media path, the media will be redirected.
-; This does not really work with in the case where Asterisk is outside and have
-; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
+; This does not really work well in the case where Asterisk is outside and the
+; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
 ;
-;canreinvite=yes                ; Asterisk by default tries to redirect the
-                                ; RTP media stream (audio) to go directly from
+;directmedia=yes                ; Asterisk by default tries to redirect the
+                                ; RTP media stream 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
+                                ; behind a NAT, or for some other reason want Asterisk to
                                 ; stay in the audio path, you may want to turn this off.
 
                                 ; This setting also affect direct RTP
@@ -684,18 +684,18 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                 ; the call directly with media peer-2-peer without re-invites.
                                 ; Will not work for video and cases where the callee sends
                                 ; RTP payloads and fmtp headers in the 200 OK that does not match the
-                                ; callers INVITE. This will also fail if canreinvite is enabled when
+                                ; callers INVITE. This will also fail if directmedia is enabled when
                                 ; the device is actually behind NAT.
 
-;canreinvite=nonat              ; An additional option is to allow media path redirection
+;directmedia=nonat              ; An additional option is to allow media path redirection
                                 ; (reinvite) but only when the peer where the media is being
                                 ; sent is known to not be behind a NAT (as the RTP core can
                                 ; determine it based on the apparent IP address the media
                                 ; arrives from).
 
-;canreinvite=update             ; Yet a third option... use UPDATE for media path redirection,
+;directmedia=update             ; Yet a third option... use UPDATE for media path redirection,
                                 ; instead of INVITE. This can be combined with 'nonat', as
-                                ; 'canreinvite=update,nonat'. It implies 'yes'.
+                                ; 'directmedia=update,nonat'. It implies 'yes'.
 
 ;ignoresdpversion=yes           ; By default, Asterisk will honor the session version
                                 ; number in SDP packets and will only modify the SDP
@@ -859,7 +859,7 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ; remotesecret
 ; transport
 ; dtmfmode
-; canreinvite
+; directmedia
 ; nat
 ; callgroup
 ; pickupgroup
@@ -969,12 +969,12 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 
 [natted-phone](!,basic-options)   ; another template inheriting basic-options
         nat=yes
-        canreinvite=no
+        directmedia=no
         host=dynamic
 
 [public-phone](!,basic-options)   ; another template inheriting basic-options
         nat=no
-        canreinvite=yes
+        directmedia=yes
 
 [my-codecs](!)                    ; a template for my preferred codecs
         disallow=all
@@ -1009,7 +1009,7 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;host=192.168.0.23               ; we have a static but private IP address
                                  ; No registration allowed
 ;nat=no                          ; there is not NAT between phone and Asterisk
-;canreinvite=yes                 ; allow RTP voice traffic to bypass Asterisk
+;directmedia=yes                 ; allow RTP voice traffic to bypass Asterisk
 ;dtmfmode=info                   ; either RFC2833 or INFO for the BudgeTone
 ;call-limit=1                    ; permit only 1 outgoing call and 1 incoming call at a time
                                  ; from the phone to asterisk (deprecated)
@@ -1039,7 +1039,7 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;callerid="Jane Smith" <5678>
 ;host=dynamic                    ; This device needs to register
 ;nat=yes                         ; X-Lite is behind a NAT router
-;canreinvite=no                  ; Typically set to NO if behind NAT
+;directmedia=no                  ; Typically set to NO if behind NAT
 ;disallow=all
 ;allow=gsm                       ; GSM consumes far less bandwidth than ulaw
 ;allow=ulaw
@@ -1112,7 +1112,7 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                  ; Send SIP and RTP to the IP address that packet is
                                  ; received from instead of trusting SIP headers
 ;host=dynamic                    ; This device registers with us
-;canreinvite=no                  ; Asterisk by default tries to redirect the
+;directmedia=no                  ; 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
diff --git a/configs/skinny.conf.sample b/configs/skinny.conf.sample
index 701723923..dfbddd4c5 100644
--- a/configs/skinny.conf.sample
+++ b/configs/skinny.conf.sample
@@ -157,7 +157,7 @@ keepalive=120
 ;device=SEP00D0BA847E6B
 ;version=P002G204	; Thanks critch
 ;context=did
-;canreinvite=yes	; Allow media to go directly between two RTP endpoints.
+;directmedia=yes	; Allow media to go directly between two RTP endpoints.
 ;line=120		; Dial(Skinny/120@florian)
 
 ; Typical config for a 7910