- Remove "incominglimit" as a configuration option in sip.conf

- Add documentation on call-limit, explaining that there's two counters
  for a type="friend".
- Document the removval of "incominglimit" in UPGRADE.txt

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7775 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/UPGRADE.txt b/UPGRADE.txt
index 27c9a0c..a0b81f8 100644 (file)
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -26,3 +26,6 @@ Variables:
   functions.  You are encouraged to move towards the associated dialplan
   function, as these variables will be removed in a future release.
 
+The SIP channel:
+
+* The "incominglimit" setting is replaced by the "call-limit" setting in sip.conf.

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index ba433e9..782a952 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -2177,8 +2177,15 @@ static void __sip_destroy(struct sip_pvt *p, int lockowner)
 }
 
 /*! \brief  update_call_counter: Handle call_limit for SIP users 
- * Note: This is going to be replaced by app_groupcount 
- * Thought: For realtime, we should propably update storage with inuse counter... */
+ * Setting a call-limit will cause calls above the limit not to be accepted.
+ *
+ * Remember that for a type=friend, there's one limit for the user and
+ * another for the peer, not a combined call limit.
+ * This will cause unexpected behaviour in subscriptions, since a "friend"
+ * is *two* devices in Asterisk, not one.
+ *
+ * Thought: For realtime, we should propably update storage with inuse counter... 
+ */
 static int update_call_counter(struct sip_pvt *fup, int event)
 {
        char name[256];
@@ -11888,7 +11895,7 @@ static struct sip_user *build_user(const char *name, struct ast_variable *v, int
                        ast_copy_string(user->musicclass, v->value, sizeof(user->musicclass));
                } else if (!strcasecmp(v->name, "accountcode")) {
                        ast_copy_string(user->accountcode, v->value, sizeof(user->accountcode));
-               } else if (!strcasecmp(v->name, "call-limit") || !strcasecmp(v->name, "incominglimit")) {
+               } else if (!strcasecmp(v->name, "call-limit")) {
                        user->call_limit = atoi(v->value);
                        if (user->call_limit < 0)
                                user->call_limit = 0;

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index e063cdc..3452b77 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -348,9 +348,15 @@ srvlookup=yes                      ; Enable DNS SRV lookups on outbound calls
 ;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
-                               ; (1 for the explicit peer, 1 for the explicit user,
+                               ; 1 for the explicit peer, 1 for the explicit user,
                                ; remember that a friend equals 1 peer and 1 user in
-                               ; memory)
+                               ; 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
+                               ; the group counters in the dial plan for that.
+                               ;
 ;mailbox=1234@default          ; mailbox 1234 in voicemail context "default"
 ;disallow=all                  ; need to disallow=all before we can use allow=
 ;allow=ulaw                    ; Note: In user sections the order of codecs