[asterisk-commits] oej: trunk r48178 - in /trunk: ./
channels/chan_sip.c configs/sip.conf.sample
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Fri Dec 1 11:16:17 MST 2006
Author: oej
Date: Fri Dec 1 12:16:16 2006
New Revision: 48178
URL: http://svn.digium.com/view/asterisk?view=rev&rev=48178
Log:
- Remove T.38 early media, since T.38 requires two way communication (imported from 1.4)
- Small fixes to limitonpeer
Modified:
trunk/ (props changed)
trunk/channels/chan_sip.c
trunk/configs/sip.conf.sample
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=48178&r1=48177&r2=48178
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Dec 1 12:16:16 2006
@@ -3099,9 +3099,9 @@
/* Check the list of users only for incoming calls */
if (global_limitonpeers == FALSE && !outgoing && (u = find_user(name, 1))) {
- inuse = &u->inUse;
- call_limit = &u->call_limit;
- inringing = NULL;
+ inuse = &u->inUse;
+ call_limit = &u->call_limit;
+ inringing = NULL;
} else if ( (p = find_peer(ast_strlen_zero(fup->peername) ? name : fup->peername, NULL, 1) ) ) { /* Try to find peer */
inuse = &p->inUse;
call_limit = &p->call_limit;
@@ -3605,15 +3605,12 @@
case AST_FRAME_MODEM:
if (p) {
sip_pvt_lock(p);
- if (p->udptl) {
- if ((ast->_state != AST_STATE_UP) &&
- !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
- !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- transmit_response_with_t38_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
- ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
- }
+ /* UDPTL requires two-way communication, so early media is not needed here.
+ we simply forget the frames if we get modem frames before the bridge is up.
+ Fax will re-transmit.
+ */
+ if (p->udptl && ast->_state != AST_STATE_UP)
res = ast_udptl_write(p->udptl, frame);
- }
sip_pvt_unlock(p);
}
break;
@@ -16268,10 +16265,10 @@
compactheaders = ast_true(v->value);
} else if (!strcasecmp(v->name, "notifymimetype")) {
ast_copy_string(default_notifymime, v->value, sizeof(default_notifymime));
+ } else if (!strcasecmp(v->name, "limitonpeers")) {
+ global_limitonpeers = ast_true(v->value);
} else if (!strcasecmp(v->name, "notifyringing")) {
global_notifyringing = ast_true(v->value);
- } else if (!strcasecmp(v->name, "limitpeersonly")) {
- global_limitonpeers = ast_true(v->value);
} else if (!strcasecmp(v->name, "notifyhold")) {
global_notifyhold = ast_true(v->value);
} else if (!strcasecmp(v->name, "alwaysauthreject")) {
Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=48178&r1=48177&r2=48178
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Fri Dec 1 12:16:16 2006
@@ -195,6 +195,12 @@
;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
+ ; 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,
+ ; one for the peer and one for the user.
;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
;
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