[asterisk-commits] trunk r37508 - in /trunk: channels/
channels/misdn/ configs/
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Thu Jul 13 07:13:24 MST 2006
Author: crichter
Date: Thu Jul 13 09:13:24 2006
New Revision: 37508
URL: http://svn.digium.com/view/asterisk?rev=37508&view=rev
Log:
added even more statefulness for sending out disconnect/release/release_complete messages. added support for incoming presentation/screening. fixed a bug that we generate TONE_EVENTS on hanguptone_indicatem, which caused asterisk to write blocking thread messages. added nodialtone option to prevent dialtone for always_immediate
Modified:
trunk/channels/chan_misdn.c
trunk/channels/misdn/chan_misdn_config.h
trunk/channels/misdn/isdn_lib.c
trunk/channels/misdn_config.c
trunk/configs/misdn.conf.sample
Modified: trunk/channels/chan_misdn.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_misdn.c?rev=37508&r1=37507&r2=37508&view=diff
==============================================================================
--- trunk/channels/chan_misdn.c (original)
+++ trunk/channels/chan_misdn.c Thu Jul 13 09:13:24 2006
@@ -2296,7 +2296,8 @@
hanguptone_indicate(p);
if (bc->nt) {
- misdn_lib_send_event( bc, EVENT_DISCONNECT);
+ if (bc->need_disconnect)
+ misdn_lib_send_event( bc, EVENT_DISCONNECT);
} else {
misdn_lib_send_event( bc, EVENT_RELEASE_COMPLETE);
p->state=MISDN_CLEANING;
@@ -2307,7 +2308,8 @@
start_bc_tones(p);
hanguptone_indicate(p);
- misdn_lib_send_event( bc, EVENT_DISCONNECT);
+ if (bc->need_disconnect)
+ misdn_lib_send_event( bc, EVENT_DISCONNECT);
break;
case MISDN_ALERTING:
@@ -2317,7 +2319,8 @@
hanguptone_indicate(p);
/*p->state=MISDN_CLEANING;*/
- misdn_lib_send_event( bc, EVENT_DISCONNECT);
+ if (bc->need_disconnect)
+ misdn_lib_send_event( bc, EVENT_DISCONNECT);
break;
case MISDN_CONNECTED:
/* Alerting or Disconect */
@@ -2326,7 +2329,8 @@
hanguptone_indicate(p);
p->bc->progress_indicator=8;
}
- misdn_lib_send_event( bc, EVENT_DISCONNECT);
+ if (bc->need_disconnect)
+ misdn_lib_send_event( bc, EVENT_DISCONNECT);
/*p->state=MISDN_CLEANING;*/
break;
@@ -2358,7 +2362,8 @@
misdn_lib_send_event(bc, EVENT_RELEASE);
p->state=MISDN_CLEANING;
} else {
- misdn_lib_send_event(bc, EVENT_DISCONNECT);
+ if (bc->need_disconnect)
+ misdn_lib_send_event(bc, EVENT_DISCONNECT);
}
}
@@ -2674,6 +2679,16 @@
{
const struct tone_zone_sound *ts= NULL;
struct ast_channel *ast=cl->ast;
+
+
+ int nd=0;
+ misdn_cfg_get( cl->bc->port, MISDN_CFG_NODIALTONE, &nd, sizeof(nd));
+
+ if (nd) {
+ chan_misdn_log(1,cl->bc->port,"Not sending Dialtone, because config wants it\n");
+ return 0;
+ }
+
chan_misdn_log(3,cl->bc->port," --> Dial\n");
ts=ast_get_indication_tone(ast->zone,"dial");
cl->ts=ts;
@@ -2692,7 +2707,6 @@
static int hanguptone_indicate(struct chan_list *cl)
{
misdn_lib_send_tone(cl->bc,TONE_HANGUP);
- misdn_lib_tone_generator_start(cl->bc);
return 0;
}
@@ -3707,12 +3721,38 @@
ch->ast->rings=1;
ast_setstate(ch->ast, AST_STATE_RINGING);
- if ( bc->pres ) {
- chan->cid.cid_pres=AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
- } else {
- chan->cid.cid_pres=AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
- }
-
+ int pres,screen;
+
+ switch (bc->pres) {
+ case 1:
+ pres=AST_PRES_RESTRICTED; chan_misdn_log(2,bc->port," --> PRES: Restricted (1)\n");
+ break;
+ case 2:
+ pres=AST_PRES_UNAVAILABLE; chan_misdn_log(2,bc->port," --> PRES: Restricted (2)\n");
+ break;
+ default:
+ pres=AST_PRES_ALLOWED; chan_misdn_log(2,bc->port," --> PRES: Restricted (%d)\n", bc->pres);
+ }
+
+ switch (bc->screen) {
+ case 0:
+ screen=AST_PRES_USER_NUMBER_UNSCREENED; chan_misdn_log(2,bc->port," --> SCREEN: Unscreened (0)\n");
+ break;
+ case 1:
+ screen=AST_PRES_USER_NUMBER_PASSED_SCREEN; chan_misdn_log(2,bc->port," --> SCREEN: Passed screen (1)\n");
+ break;
+ case 2:
+ screen=AST_PRES_USER_NUMBER_FAILED_SCREEN; chan_misdn_log(2,bc->port," --> SCREEN: failed screen (2)\n");
+ break;
+ case 3:
+ screen=AST_PRES_NETWORK_NUMBER; chan_misdn_log(2,bc->port," --> SCREEN: Network Number (3)\n");
+ break;
+ default:
+ screen=AST_PRES_USER_NUMBER_UNSCREENED; chan_misdn_log(2,bc->port," --> SCREEN: Unscreened (%d)\n",bc->screen);
+ }
+
+ chan->cid.cid_pres=pres+screen;
+
pbx_builtin_setvar_helper(chan, "TRANSFERCAPABILITY", ast_transfercapability2str(bc->capability));
chan->transfercapability=bc->capability;
Modified: trunk/channels/misdn/chan_misdn_config.h
URL: http://svn.digium.com/view/asterisk/trunk/channels/misdn/chan_misdn_config.h?rev=37508&r1=37507&r2=37508&view=diff
==============================================================================
--- trunk/channels/misdn/chan_misdn_config.h (original)
+++ trunk/channels/misdn/chan_misdn_config.h Thu Jul 13 09:13:24 2006
@@ -44,6 +44,7 @@
MISDN_CFG_PRES, /* int */
MISDN_CFG_SCREEN, /* int */
MISDN_CFG_ALWAYS_IMMEDIATE, /* int (bool) */
+ MISDN_CFG_NODIALTONE, /* int (bool) */
MISDN_CFG_IMMEDIATE, /* int (bool) */
MISDN_CFG_SENDDTMF, /* int (bool) */
MISDN_CFG_HOLD_ALLOWED, /* int (bool) */
Modified: trunk/channels/misdn/isdn_lib.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/misdn/isdn_lib.c?rev=37508&r1=37507&r2=37508&view=diff
==============================================================================
--- trunk/channels/misdn/isdn_lib.c (original)
+++ trunk/channels/misdn/isdn_lib.c Thu Jul 13 09:13:24 2006
@@ -3195,6 +3195,7 @@
cb_log(-1,bc->port," --> we have already send Release\n");
return -1;
}
+ bc->need_disconnect=0;
bc->need_release=0;
break;
case EVENT_RELEASE_COMPLETE:
@@ -3202,6 +3203,8 @@
cb_log(-1,bc->port," --> we have already send Release_complete\n");
return -1;
}
+ bc->need_disconnect=0;
+ bc->need_release=0;
bc->need_release_complete=0;
break;
Modified: trunk/channels/misdn_config.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/misdn_config.c?rev=37508&r1=37507&r2=37508&view=diff
==============================================================================
--- trunk/channels/misdn_config.c (original)
+++ trunk/channels/misdn_config.c Thu Jul 13 09:13:24 2006
@@ -210,6 +210,10 @@
"\tThere you can use DigitTimeout if you can't or don't want to use\n"
"\tisdn overlap dial.\n"
"\tNOTE: This will jump into the s extension for every exten!" },
+ { "nodialtone", MISDN_CFG_NODIALTONE, MISDN_CTYPE_BOOL, "no", NONE,
+ "Enable this to prevent chan_misdn to generate the dialtone\n"
+ "\tThis makes only sense together with the always_immediate=yes option\n"
+ "\tto generate your own dialtone with Playtones or so.\n"},
{ "immediate", MISDN_CFG_IMMEDIATE, MISDN_CTYPE_BOOL, "no", NONE,
"Enable this if you want callers which called exactly the base\n"
"\tnumber (so no extension is set) to jump into the s extension.\n"
Modified: trunk/configs/misdn.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/misdn.conf.sample?rev=37508&r1=37507&r2=37508&view=diff
==============================================================================
--- trunk/configs/misdn.conf.sample (original)
+++ trunk/configs/misdn.conf.sample Thu Jul 13 09:13:24 2006
@@ -272,6 +272,15 @@
;
;always_immediate=no
+;
+; set this to yes if you want to generate your own dialtone
+; with always_immediate=yes, else chan_misdn generates the dialtone
+;
+; default value: no
+;
+nodialtone=no
+
+
; uncomment the following if you want callers which called exactly the
; base number (so no extension is set) jump to the s extension.
; if the user dials something more it jumps to the correct extension
More information about the asterisk-commits
mailing list