- Deprecate "call-limit" in chan_sip. No other channel driver enforces call-limits

  and we now have the groupcount system to implement call-limits in the dialplan. You
  can use the "setvar" option in realtime/sip.conf to set limits per device.

- Implement "callcounter" as a new option to enable the call counting we need to
  report device status to queue, manager and SIP subscriptions.

The call counter setting is now enabled in the code by setting the device call-limit
to 999. When we remove the call limit, we can simply enable this with a boolean
setting.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89554 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 21 insertions, 15 deletions

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index ae04d37bd..ae5c9ba61 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -23,6 +23,15 @@
 ;				Active SIP peers will not be reconfigured
 ;
 
+; ** Deprecated options **
+; The "call-limit" configuation option is deprecated. It still works in
+; this version of Asterisk, but will disappear in the next version.
+; You are encouraged to use the dialplan groupcount functionality
+; to enforce call limits instead of using this channel-specific method.
+;
+; You can still set limits per device in sip.conf or in a database by using 
+; "setvar" to set variables that can be used in the dialplan for various limits.
+
 [general]
 context=default			; Default context for incoming calls
 ;allowguest=no			; Allow or reject guest calls (default is yes)
@@ -206,14 +215,11 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 ; (See extensions.conf.sample for examples)
 ; chan_sip support two major formats for notifications: dialog-info and SIMPLE 
 ;
-; You will get more detailed reports (busy etc) if you have a call limit set
-; for a device. When the call limit is filled, we will indicate busy. Note that
-; you need at least 2 in order to be able to do attended transfers.
+; You will get more detailed reports (busy etc) if you have a call counter enabled
+; for a device. 
 ;
-; If you set the busylevel in addition to the call limit, we will indicate busy
-; when we have a number of calls that matches busylevel, but still allow calls
-; up to the call-limit. This allows for transfers while still having blinking
-; lamps and queues understanding that a device is busy.
+; If you set the busylevel, we will indicate busy when we have a number of calls that 
+; matches the busylevel treshold.
 ;
 ; For queues, you will need this level of detail in status reporting, regardless
 ; if you use SIP subscriptions. Queues and manager use the same internal interface
@@ -230,12 +236,14 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 ;notifyhold = yes		; Notify subscriptions on HOLD state (default: no)
 				; Turning on notifyringing and notifyhold will add a lot
 				; more database transactions if you are using realtime.
-;limitonpeer = yes		; Apply call limits on peers only. This will improve 
+;limitonpeer = yes		; Apply call counting on peers only. This will improve 
 				; status notification when you are using type=friend
 				; Inbound calls, that really apply to the user part
 				; of a friend will now be added to and compared with
-				; the peer limit instead of applying two call limits,
+				; the peer counter instead of applying two call counters,
 				; one for the peer and one for the user.
+;callcounter = yes		; Enable call counters on devices. This can be set per
+				; device too.
 
 ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
 ;
@@ -561,7 +569,8 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 ; setvar                      setvar
 ; callerid		      callerid
 ; amaflags		      amaflags
-; call-limit		      call-limit
+; call-limit		      call-limit	(deprecated)
+; callcounter                 callcounter
 ; allowoverlap		      allowoverlap
 ; allowsubscribe	      allowsubscribe
 ; allowtransfer	      	      allowtransfer
@@ -603,9 +612,7 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 ;fromdomain=provider.sip.domain	
 ;host=box.provider.com
 ;usereqphone=yes			; This provider requires ";user=phone" on URI
-;call-limit=5				; permit only 5 simultaneous outgoing calls to this peer
-					; Call-limits will not be enforced on real-time peers,
-					; since they are not stored in-memory
+;callcounter=yes			; Enable call counter
 ;busylevel=2				; Signal busy at 2 or more calls
 ;outboundproxy=proxy.provider.domain	; send outbound signaling to this proxy, not directly to the peer
 ;port=80				; The port number we want to connect to on the remote side
@@ -690,11 +697,10 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 ;canreinvite=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
+				; from the phone to asterisk (deprecated)
 				; 1 for the explicit peer, 1 for the explicit user,
 				; remember that a friend equals 1 peer and 1 user in
 				; memory
-				; This will affect your subscriptions as well.
 				; There is no combined call counter for a "friend"
 				; so there's currently no way in sip.conf to limit
 				; to one inbound or outbound call per phone. Use