[Asterisk-Users] Is a BUG ? Hints and incominglimit
Alvaro Parres
aparres at gmail.com
Mon Dec 5 17:30:29 MST 2005
which version of Asterisk do you have ?, Becouse when i change the function
to your code, every time that one phone with call-limit the Asterisk crash.
I have 1.2.0
On 12/3/05, Paradise Dove <pardove at gmail.com> wrote:
>
> hi,
> This is the new update_call_counter() which works fine for me:
>
> /*! \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... */
> static int update_call_counter(struct sip_pvt *fup, int event)
> {
> char name[256];
> int *inuse, *call_limit;
> int outgoing = ast_test_flag(fup, SIP_OUTGOING);
> struct sip_user *u = NULL;
> struct sip_peer *p = NULL;
>
> if (option_debug > 2)
> ast_log(LOG_DEBUG, "Updating call counter for %s call\n",
> outgoing ? "outgoing" : "incoming");
> /* Test if we need to check call limits, in order to avoid
> realtime lookups if we do not need it */
> if (!ast_test_flag(fup, SIP_CALL_LIMIT))
> return 0;
>
> ast_copy_string(name, fup->username, sizeof(name));
>
> /* Check the list of users */
> // paradise dove
> p = find_peer(name, NULL, 1);
> if (p) {
> inuse = &p->inUse;
> call_limit = &p->call_limit;
> } else if (!u) {
> /* Try to find user */
> u = find_user(name, 1);
> if (u) {
> inuse = &u->inUse;
> call_limit = &u->call_limit;
> } else {
> if (option_debug > 1)
> ast_log(LOG_DEBUG, "%s is not a local user, no call
> limit\n", name);
> return 0;
> }
> }
> switch(event) {
> /* incoming and outgoing affects the inUse counter */
> case DEC_CALL_LIMIT:
> if ( *inuse > 0 ) {
> (*inuse)--;
> } else {
> *inuse = 0;
> }
> if (option_debug > 1 || sipdebug) {
> ast_log(LOG_DEBUG, "Call %s %s '%s' removed from call
> limit %d\n", outgoing ? "to" : "from", u ? "user":"peer"
> }
> break;
> case INC_CALL_LIMIT:
> if (*call_limit > 0 ) {
> if (*inuse >= *call_limit) {
> ast_log(LOG_ERROR, "Call %s %s '%s' rejected due
> to usage limit of %d\n", outgoing ? "to" : "from", u ? "u
> // paradise dove
> if (p)
> ASTOBJ_UNREF(p,sip_destroy_peer);
> else if (u)
> ASTOBJ_UNREF(u,sip_destroy_user);
> return -1;
> }
> }
> (*inuse)++;
> if (option_debug > 1 || sipdebug) {
> ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of
> %d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *in
> }
> break;
> default:
> ast_log(LOG_ERROR, "update_call_counter(%s, %d) called
> with no event!\n", name, event);
> }
> // paradise dove
> if (p)
> ASTOBJ_UNREF(p,sip_destroy_peer);
> else if (u)
> ASTOBJ_UNREF(u,sip_destroy_user);
> return 0;
> }
>
> Paradise Dove
>
>
> On 12/2/05, Alvaro Parres <aparres at gmail.com> wrote:
> > Could you send it patch please.
> >
> >
> >
> >
> > On 11/30/05, Paradise Dove <pardove at gmail.com> wrote:
> > >
> > > btw, i've patched this part of code and now its working fine for me.
> > > i'm going to upload it.
> > >
> > > Paradise Dove
> > >
> > > On 11/30/05, Kevin Hanson <tuxpert at comcast.net> wrote:
> > > > Paradise Dove wrote:
> > > >
> > > > >>Yes with version 1.2. I have tried already with call-limit and the
> > same.
> > > > >>
> > > > >>
> > > > >i agree with you, it seems to be a bug which i've submited before
> (bug
> > > > >#5281) but it's now closed by bug marshals!!!!!
> > > > >
> > > > >
> > > > >
> > > > It's not closed. It's suspended waiting input from you:
> > > >
> > > > "Closing until the appropriate debug/trace output can be provided."
> > > >
> > > > On 10/30 you said you were still trying to get the debug output.
> > > >
> > > > Cheers,
> > > > Kevin
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