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

Browse Source 
- 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
This commit is contained in:
Olle Johansson
2006-01-04 09:10:56 +00:00
parent 1d603f5256
commit 5462ec082c
3 changed files with 21 additions and 5 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
UPGRADE.txt
View File

@@ -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.

13
channels/chan_sip.c
View File

@@ -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;

10
configs/sip.conf.sample
View File

@@ -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