[svn-commits] branch kpfleming/stringfields - r7791 in
/team/kpfleming/stringfields: ./ app...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Wed Jan 4 14:40:09 CST 2006
Author: kpfleming
Date: Wed Jan 4 14:40:08 2006
New Revision: 7791
URL: http://svn.digium.com/view/asterisk?rev=7791&view=rev
Log:
update to current trunk
Modified:
team/kpfleming/stringfields/ (props changed)
team/kpfleming/stringfields/UPGRADE.txt
team/kpfleming/stringfields/apps/app_disa.c
team/kpfleming/stringfields/apps/app_privacy.c
team/kpfleming/stringfields/cdr.c
team/kpfleming/stringfields/channel.c
team/kpfleming/stringfields/channels/chan_sip.c
team/kpfleming/stringfields/channels/chan_zap.c
team/kpfleming/stringfields/configs/sip.conf.sample
team/kpfleming/stringfields/contrib/scripts/managerproxy.pl
team/kpfleming/stringfields/funcs/func_moh.c
team/kpfleming/stringfields/res/res_osp.c
Propchange: team/kpfleming/stringfields/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Wed Jan 4 14:40:08 2006
@@ -1,1 +1,1 @@
-/trunk:1-7751
+/trunk:1-7790
Modified: team/kpfleming/stringfields/UPGRADE.txt
URL: http://svn.digium.com/view/asterisk/team/kpfleming/stringfields/UPGRADE.txt?rev=7791&r1=7790&r2=7791&view=diff
==============================================================================
--- team/kpfleming/stringfields/UPGRADE.txt (original)
+++ team/kpfleming/stringfields/UPGRADE.txt Wed Jan 4 14:40:08 2006
@@ -26,3 +26,6 @@
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.
Modified: team/kpfleming/stringfields/apps/app_disa.c
URL: http://svn.digium.com/view/asterisk/team/kpfleming/stringfields/apps/app_disa.c?rev=7791&r1=7790&r2=7791&view=diff
==============================================================================
--- team/kpfleming/stringfields/apps/app_disa.c (original)
+++ team/kpfleming/stringfields/apps/app_disa.c Wed Jan 4 14:40:08 2006
@@ -146,12 +146,12 @@
}
if (ast_set_write_format(chan,AST_FORMAT_ULAW)) {
- ast_log(LOG_WARNING, "Unable to set write format to Mu-law on %s\n",chan->name);
+ ast_log(LOG_WARNING, "Unable to set write format to Mu-law on %s\n", chan->name);
LOCAL_USER_REMOVE(u);
return -1;
}
if (ast_set_read_format(chan,AST_FORMAT_ULAW)) {
- ast_log(LOG_WARNING, "Unable to set read format to Mu-law on %s\n",chan->name);
+ ast_log(LOG_WARNING, "Unable to set read format to Mu-law on %s\n", chan->name);
LOCAL_USER_REMOVE(u);
return -1;
}
@@ -198,8 +198,7 @@
for (;;) {
/* if outa time, give em reorder */
if (ast_tvdiff_ms(ast_tvnow(), lastdigittime) >
- ((k&2) ? digittimeout : firstdigittimeout))
- {
+ ((k&2) ? digittimeout : firstdigittimeout)) {
ast_log(LOG_DEBUG,"DISA %s entry timeout on chan %s\n",
((k&1) ? "extension" : "password"),chan->name);
break;
@@ -210,14 +209,12 @@
}
f = ast_read(chan);
- if (f == NULL)
- {
+ if (f == NULL) {
LOCAL_USER_REMOVE(u);
return -1;
}
if ((f->frametype == AST_FRAME_CONTROL) &&
- (f->subclass == AST_CONTROL_HANGUP))
- {
+ (f->subclass == AST_CONTROL_HANGUP)) {
ast_frfree(f);
LOCAL_USER_REMOVE(u);
return -1;
@@ -226,56 +223,54 @@
ast_frfree(f);
continue;
}
- /* if not DTMF, just do it again */
- if (f->frametype != AST_FRAME_DTMF)
- {
+
+ /* if not DTMF, just do it again */
+ if (f->frametype != AST_FRAME_DTMF) {
ast_frfree(f);
continue;
}
j = f->subclass; /* save digit */
ast_frfree(f);
- if (i == 0)
- {
+ if (i == 0) {
k|=2; /* We have the first digit */
ast_playtones_stop(chan);
}
lastdigittime = ast_tvnow();
/* got a DTMF tone */
- if (i < AST_MAX_EXTENSION) /* if still valid number of digits */
- {
- if (!(k&1)) /* if in password state */
- {
- if (j == '#') /* end of password */
- {
+ if (i < AST_MAX_EXTENSION) { /* if still valid number of digits */
+ if (!(k&1)) { /* if in password state */
+ if (j == '#') { /* end of password */
/* see if this is an integer */
- if (sscanf(args.passcode,"%d",&j) < 1)
- { /* nope, it must be a filename */
+ if (sscanf(args.passcode,"%d",&j) < 1) { /* nope, it must be a filename */
fp = fopen(args.passcode,"r");
- if (!fp)
- {
+ if (!fp) {
ast_log(LOG_WARNING,"DISA password file %s not found on chan %s\n",args.passcode,chan->name);
LOCAL_USER_REMOVE(u);
return -1;
- }
+ }
pwline[0] = 0;
- while(fgets(pwline,sizeof(pwline) - 1,fp))
- {
- if (!pwline[0]) continue;
+ while(fgets(pwline,sizeof(pwline) - 1,fp)) {
+ if (!pwline[0])
+ continue;
if (pwline[strlen(pwline) - 1] == '\n')
pwline[strlen(pwline) - 1] = 0;
- if (!pwline[0]) continue;
- /* skip comments */
- if (pwline[0] == '#') continue;
- if (pwline[0] == ';') continue;
+ if (!pwline[0])
+ continue;
+ /* skip comments */
+ if (pwline[0] == '#')
+ continue;
+ if (pwline[0] == ';')
+ continue;
AST_STANDARD_APP_ARGS(args, pwline);
ast_log(LOG_DEBUG, "Mailbox: %s\n",args.mailbox);
- /* password must be in valid format (numeric) */
- if (sscanf(args.passcode,"%d",&j) < 1) continue;
- /* if we got it */
+ /* password must be in valid format (numeric) */
+ if (sscanf(args.passcode,"%d", &j) < 1)
+ continue;
+ /* if we got it */
if (!strcmp(exten,args.passcode)) {
if (ast_strlen_zero(args.context))
args.context = "disa";
@@ -283,12 +278,11 @@
args.mailbox = "";
break;
}
- }
+ }
fclose(fp);
- }
- /* compare the two */
- if (strcmp(exten,args.passcode))
- {
+ }
+ /* compare the two */
+ if (strcmp(exten,args.passcode)) {
ast_log(LOG_WARNING,"DISA on chan %s got bad password %s\n",chan->name,exten);
goto reorder;
@@ -302,15 +296,16 @@
exten[sizeof(acctcode)] = 0;
ast_copy_string(acctcode, exten, sizeof(acctcode));
exten[0] = 0;
- ast_log(LOG_DEBUG,"Successful DISA log-in on chan %s\n",chan->name);
+ ast_log(LOG_DEBUG,"Successful DISA log-in on chan %s\n", chan->name);
continue;
}
}
exten[i++] = j; /* save digit */
exten[i] = 0;
- if (!(k&1)) continue; /* if getting password, continue doing it */
- /* if this exists */
+ if (!(k&1))
+ continue; /* if getting password, continue doing it */
+ /* if this exists */
if (ast_ignore_pattern(args.context, exten)) {
play_dialtone(chan, "");
@@ -321,7 +316,7 @@
did_ignore = 0;
}
- /* if can do some more, do it */
+ /* if can do some more, do it */
if (!ast_matchmore_extension(chan,args.context,exten,1, chan->cid.cid_num)) {
break;
}
@@ -341,8 +336,7 @@
if (!recheck || ast_exists_extension(chan, args.context, exten, 1, chan->cid.cid_num)) {
ast_playtones_stop(chan);
/* We're authenticated and have a target extension */
- if (!ast_strlen_zero(args.cid))
- {
+ if (!ast_strlen_zero(args.cid)) {
ast_callerid_split(args.cid, ourcidname, sizeof(ourcidname), ourcidnum, sizeof(ourcidnum));
ast_set_callerid(chan, ourcidnum, ourcidname, ourcidnum);
}
@@ -364,8 +358,7 @@
ast_indicate(chan,AST_CONTROL_CONGESTION);
/* something is invalid, give em reorder for several seconds */
time(&rstart);
- while(time(NULL) < rstart + 10)
- {
+ while(time(NULL) < rstart + 10) {
if (ast_waitfor(chan, -1) < 0)
break;
f = ast_read(chan);
Modified: team/kpfleming/stringfields/apps/app_privacy.c
URL: http://svn.digium.com/view/asterisk/team/kpfleming/stringfields/apps/app_privacy.c?rev=7791&r1=7790&r2=7791&view=diff
==============================================================================
--- team/kpfleming/stringfields/apps/app_privacy.c (original)
+++ team/kpfleming/stringfields/apps/app_privacy.c Wed Jan 4 14:40:08 2006
@@ -195,7 +195,7 @@
}
/*Got a number, play sounds and send them on their way*/
- if ((retries < maxretries) && res == 1 ) {
+ if ((retries < maxretries) && !res ) {
res = ast_streamfile(chan, "privacy-thankyou", chan->language);
if (!res)
res = ast_waitstream(chan, "");
Modified: team/kpfleming/stringfields/cdr.c
URL: http://svn.digium.com/view/asterisk/team/kpfleming/stringfields/cdr.c?rev=7791&r1=7790&r2=7791&view=diff
==============================================================================
--- team/kpfleming/stringfields/cdr.c (original)
+++ team/kpfleming/stringfields/cdr.c Wed Jan 4 14:40:08 2006
@@ -22,7 +22,7 @@
*
* \author Mark Spencer <markster at digium.com>
*
- * Includes code and algorithms from the Zapata library.
+ * \note Includes code and algorithms from the Zapata library.
*
* \note We do a lot of checking here in the CDR code to try to be sure we don't ever let a CDR slip
* through our fingers somehow. If someone allocates a CDR, it must be completely handled normally
Modified: team/kpfleming/stringfields/channel.c
URL: http://svn.digium.com/view/asterisk/team/kpfleming/stringfields/channel.c?rev=7791&r1=7790&r2=7791&view=diff
==============================================================================
--- team/kpfleming/stringfields/channel.c (original)
+++ team/kpfleming/stringfields/channel.c Wed Jan 4 14:40:08 2006
@@ -168,8 +168,10 @@
static int show_channeltypes(int fd, int argc, char *argv[])
{
-#define FORMAT "%-10.10s %-30.30s %-12.12s %-12.12s %-12.12s\n"
+#define FORMAT "%-10.10s %-40.40s %-12.12s %-12.12s %-12.12s\n"
struct chanlist *cl;
+ int count_chan = 0;
+
ast_cli(fd, FORMAT, "Type", "Description", "Devicestate", "Indications", "Transfer");
ast_cli(fd, FORMAT, "----------", "-----------", "-----------", "-----------", "--------");
if (ast_mutex_lock(&chlock)) {
@@ -181,8 +183,10 @@
(cl->tech->devicestate) ? "yes" : "no",
(cl->tech->indicate) ? "yes" : "no",
(cl->tech->transfer) ? "yes" : "no");
+ count_chan++;
}
ast_mutex_unlock(&chlock);
+ ast_cli(fd, "----------\n%d channel drivers registered.\n", count_chan);
return RESULT_SUCCESS;
#undef FORMAT
Modified: team/kpfleming/stringfields/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/kpfleming/stringfields/channels/chan_sip.c?rev=7791&r1=7790&r2=7791&view=diff
==============================================================================
--- team/kpfleming/stringfields/channels/chan_sip.c (original)
+++ team/kpfleming/stringfields/channels/chan_sip.c Wed Jan 4 14:40:08 2006
@@ -2180,8 +2180,15 @@
}
/*! \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];
@@ -2276,21 +2283,54 @@
/* Possible values taken from causes.h */
switch(cause) {
- case 603: /* Declined */
+ case 401: /* Unauthorized */
+ return AST_CAUSE_CALL_REJECTED;
case 403: /* Not found */
return AST_CAUSE_CALL_REJECTED;
case 404: /* Not found */
return AST_CAUSE_UNALLOCATED;
+ case 405: /* Method not allowed */
+ return AST_CAUSE_INTERWORKING;
+ case 407: /* Proxy authentication required */
+ return AST_CAUSE_CALL_REJECTED;
case 408: /* No reaction */
return AST_CAUSE_NO_USER_RESPONSE;
+ case 409: /* Conflict */
+ return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
+ case 410: /* Gone */
+ return AST_CAUSE_UNALLOCATED;
+ case 411: /* Length required */
+ return AST_CAUSE_INTERWORKING;
+ case 413: /* Request entity too large */
+ return AST_CAUSE_INTERWORKING;
+ case 414: /* Request URI too large */
+ return AST_CAUSE_INTERWORKING;
+ case 415: /* Unsupported media type */
+ return AST_CAUSE_INTERWORKING;
+ case 420: /* Bad extension */
+ return AST_CAUSE_NO_ROUTE_DESTINATION;
case 480: /* No answer */
return AST_CAUSE_FAILURE;
+ case 481: /* No answer */
+ return AST_CAUSE_INTERWORKING;
+ case 482: /* Loop detected */
+ return AST_CAUSE_INTERWORKING;
case 483: /* Too many hops */
return AST_CAUSE_NO_ANSWER;
+ case 484: /* Address incomplete */
+ return AST_CAUSE_INVALID_NUMBER_FORMAT;
+ case 485: /* Ambigous */
+ return AST_CAUSE_UNALLOCATED;
case 486: /* Busy everywhere */
return AST_CAUSE_BUSY;
+ case 487: /* Request terminated */
+ return AST_CAUSE_INTERWORKING;
case 488: /* No codecs approved */
return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
+ case 491: /* Request pending */
+ return AST_CAUSE_INTERWORKING;
+ case 493: /* Undecipherable */
+ return AST_CAUSE_INTERWORKING;
case 500: /* Server internal failure */
return AST_CAUSE_FAILURE;
case 501: /* Call rejected */
@@ -2299,6 +2339,18 @@
return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
case 503: /* Service unavailable */
return AST_CAUSE_CONGESTION;
+ case 504: /* Gateway timeout */
+ return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
+ case 505: /* SIP version not supported */
+ return AST_CAUSE_INTERWORKING;
+ case 600: /* Busy everywhere */
+ return AST_CAUSE_USER_BUSY;
+ case 603: /* Decline */
+ return AST_CAUSE_CALL_REJECTED;
+ case 604: /* Does not exist anywhere */
+ return AST_CAUSE_UNALLOCATED;
+ case 606: /* Not acceptable */
+ return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
default:
return AST_CAUSE_NORMAL;
}
@@ -4274,7 +4326,7 @@
" <media_control>\r\n"
" <vc_primitive>\r\n"
" <to_encoder>\r\n"
- " <picture_fast_update\r\n"
+ " <picture_fast_update>\r\n"
" </picture_fast_update>\r\n"
" </to_encoder>\r\n"
" </vc_primitive>\r\n"
@@ -11930,7 +11982,7 @@
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;
Modified: team/kpfleming/stringfields/channels/chan_zap.c
URL: http://svn.digium.com/view/asterisk/team/kpfleming/stringfields/channels/chan_zap.c?rev=7791&r1=7790&r2=7791&view=diff
==============================================================================
--- team/kpfleming/stringfields/channels/chan_zap.c (original)
+++ team/kpfleming/stringfields/channels/chan_zap.c Wed Jan 4 14:40:08 2006
@@ -7972,13 +7972,13 @@
}
if ((dchan >= 0) && (span >= 0)) {
if (dchancount > 1)
- ast_log(LOG_DEBUG, "[Span %d D-Channel %d]%s", span, dchan, s);
+ ast_verbose("[Span %d D-Channel %d]%s", span, dchan, s);
else
- ast_log(LOG_DEBUG, "%s", s);
+ ast_verbose("%s", s);
} else
ast_log(LOG_ERROR, "PRI debug error: could not find pri associated it with debug message output\n");
} else
- ast_log(LOG_DEBUG, "%s", s);
+ ast_verbose("%s", s);
ast_mutex_lock(&pridebugfdlock);
Modified: team/kpfleming/stringfields/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/team/kpfleming/stringfields/configs/sip.conf.sample?rev=7791&r1=7790&r2=7791&view=diff
==============================================================================
--- team/kpfleming/stringfields/configs/sip.conf.sample (original)
+++ team/kpfleming/stringfields/configs/sip.conf.sample Wed Jan 4 14:40:08 2006
@@ -348,9 +348,15 @@
;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 at 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
Modified: team/kpfleming/stringfields/contrib/scripts/managerproxy.pl
URL: http://svn.digium.com/view/asterisk/team/kpfleming/stringfields/contrib/scripts/managerproxy.pl?rev=7791&r1=7790&r2=7791&view=diff
==============================================================================
--- team/kpfleming/stringfields/contrib/scripts/managerproxy.pl (original)
+++ team/kpfleming/stringfields/contrib/scripts/managerproxy.pl Wed Jan 4 14:40:08 2006
@@ -2,14 +2,14 @@
#
# Simple Asterisk Manager Proxy, Version 1.01
# 2004-09-26
-# Copyright (c) 2004 David C. Troy <dave at popvox.com>
+# Copyright (c) 2004 David C. Troy <dave at popvox.com>
#
# This code is based on Flash Operator Panel 'op_server.pl'
-# by Nicolá³ Gudiñ¯¬
+# by Nicolas Gudino
# Copyright (C) 2004.
#
-# David C. Troy <dave at popvox.com>
-# Nicolá³ Gudiñ¯ ¼nicolas at house.com.ar>
+# David C. Troy <dave at popvox.com>
+# Nicolas Gudino <nicolas at house.com.ar>
#
# This program is free software, distributed under the terms of
# the GNU General Public License.
Modified: team/kpfleming/stringfields/funcs/func_moh.c
URL: http://svn.digium.com/view/asterisk/team/kpfleming/stringfields/funcs/func_moh.c?rev=7791&r1=7790&r2=7791&view=diff
==============================================================================
--- team/kpfleming/stringfields/funcs/func_moh.c (original)
+++ team/kpfleming/stringfields/funcs/func_moh.c Wed Jan 4 14:40:08 2006
@@ -40,7 +40,7 @@
static void function_moh_write(struct ast_channel *chan, char *cmd, char *data, const char *value)
{
- ast_copy_string(chan->musicclass, value, MAX_MUSICCLASS);
+ ast_copy_string(chan->musicclass, value, sizeof(chan->musicclass));
}
#ifndef BUILTIN_FUNC
Modified: team/kpfleming/stringfields/res/res_osp.c
URL: http://svn.digium.com/view/asterisk/team/kpfleming/stringfields/res/res_osp.c?rev=7791&r1=7790&r2=7791&view=diff
==============================================================================
--- team/kpfleming/stringfields/res/res_osp.c (original)
+++ team/kpfleming/stringfields/res/res_osp.c Wed Jan 4 14:40:08 2006
@@ -516,6 +516,44 @@
return res;
}
+static int check_dest(struct ast_osp_result *result, char *token, int tokensize)
+{
+ OSPE_DEST_OSP_ENABLED enabled;
+ OSPE_DEST_PROT prot;
+ int res = 1;
+
+ /* Check destination OSP version */
+ if (!OSPPTransactionIsDestOSPEnabled(result->handle, &enabled) && (enabled == OSPE_OSP_FALSE)) {
+ result->token[0] = 0;
+ } else {
+ ast_base64encode(result->token, token, tokensize, sizeof(result->token) - 1);
+ }
+
+ /* Check destination protocol */
+ if (OSPPTransactionGetDestProtocol(result->handle, &prot)) {
+ prot = OSPE_DEST_PROT_UNDEFINED;
+ }
+ switch(prot) {
+ case OSPE_DEST_PROT_UNDEFINED: /* Protocol is not configured, use SIP as default */
+ case OSPE_DEST_PROT_SIP:
+ ast_copy_string(result->tech, "SIP", sizeof(result->tech));
+ break;
+ case OSPE_DEST_PROT_H323_SETUP:
+ ast_copy_string(result->tech, "H323", sizeof(result->tech));
+ break;
+ case OSPE_DEST_PROT_IAX:
+ ast_copy_string(result->tech, "IAX", sizeof(result->tech));
+ break;
+ case OSPE_DEST_PROT_H323_LRQ:
+ case OSPE_DEST_PROT_UNKNOWN:
+ default:
+ ast_log(LOG_DEBUG, "Unknown destination protocol '%d', skipping...\n", prot);
+ res = 0;
+ }
+
+ return res;
+}
+
int ast_osp_lookup(struct ast_channel *chan, char *provider, char *extension, char *callerid, struct ast_osp_result *result)
{
int cres;
@@ -533,8 +571,6 @@
char destination[2048]="";
char token[2000];
char tmp[256]="", *l, *n;
- OSPE_DEST_PROT prot;
- OSPE_DEST_OSP_ENABLED ospenabled;
char *devinfo = NULL;
result->handle = -1;
@@ -603,49 +639,27 @@
ast_channel_setwhentohangup (chan, timelimit);
}
do {
- if (!OSPPTransactionIsDestOSPEnabled (result->handle, &ospenabled) && (ospenabled == OSPE_OSP_FALSE)) {
- result->token[0] = 0;
- }
- else {
- ast_base64encode(result->token, token, tokenlen, sizeof(result->token) - 1);
- }
- if ((strlen(destination) > 2) && !OSPPTransactionGetDestProtocol(result->handle, &prot)) {
- res = 1;
+ if ((strlen(destination) > 2) && (check_dest(result, token, tokenlen))) {
/* Strip leading and trailing brackets */
destination[strlen(destination) - 1] = '\0';
- switch(prot) {
- case OSPE_DEST_PROT_H323_SETUP:
- ast_copy_string(result->tech, "H323", sizeof(result->tech));
- snprintf(result->dest, sizeof(result->dest), "%s@%s", callednum, destination + 1);
- break;
- case OSPE_DEST_PROT_SIP:
- ast_copy_string(result->tech, "SIP", sizeof(result->tech));
- snprintf(result->dest, sizeof(result->dest), "%s@%s", callednum, destination + 1);
- break;
- case OSPE_DEST_PROT_IAX:
- ast_copy_string(result->tech, "IAX", sizeof(result->tech));
- snprintf(result->dest, sizeof(result->dest), "%s@%s", callednum, destination + 1);
- break;
- default:
- ast_log(LOG_DEBUG, "Unknown destination protocol '%d', skipping...\n", prot);
- res = 0;
- }
- if (!res && result->numresults) {
+ snprintf(result->dest, sizeof(result->dest), "%s@%s", callednum, destination + 1);
+ res = 1;
+ } else {
+ if(result->numresults) {
result->numresults--;
callidlen = sizeof(callidstr);
- if (OSPPTransactionGetNextDestination(result->handle, OSPC_FAIL_INCOMPATIBLE_DEST, 0, NULL, NULL, &timelimit, &callidlen, callidstr,
- sizeof(callednum), callednum, sizeof(callingnum), callingnum, sizeof(destination), destination, 0, NULL, &tokenlen, token)) {
- break;
+ if (!OSPPTransactionGetNextDestination(result->handle, OSPC_FAIL_INCOMPATIBLE_DEST, 0, NULL, NULL, &timelimit, &callidlen,
+ callidstr, sizeof(callednum), callednum, sizeof(callingnum), callingnum, sizeof(destination), destination, 0, NULL,
+ &tokenlen, token))
+ {
+ continue;
}
}
- } else {
- ast_log(LOG_DEBUG, "Missing destination protocol\n");
- break;
}
- } while(!res && result->numresults);
+ break;
+ } while (1);
}
}
-
}
if (!res) {
OSPPTransactionDelete(result->handle);
@@ -667,7 +681,6 @@
{
int res = 0;
int tokenlen;
- unsigned int dummy=0;
unsigned int timelimit;
unsigned int callidlen;
char callidstr[OSPC_CALLID_MAXSIZE] = "";
@@ -675,62 +688,33 @@
char callingnum[2048]="";
char destination[2048]="";
char token[2000];
- OSPE_DEST_PROT prot;
- OSPE_DEST_OSP_ENABLED ospenabled;
result->tech[0] = '\0';
result->dest[0] = '\0';
result->token[0] = '\0';
if (result->handle > -1) {
- dummy = 0;
if (result->numresults) {
tokenlen = sizeof(token);
while(!res && result->numresults) {
result->numresults--;
callidlen = sizeof(callidstr);
if (!OSPPTransactionGetNextDestination(result->handle, OSPC_FAIL_INCOMPATIBLE_DEST, 0, NULL, NULL, &timelimit, &callidlen, callidstr,
- sizeof(callednum), callednum, sizeof(callingnum), callingnum, sizeof(destination), destination, 0, NULL, &tokenlen, token)) {
- if (!OSPPTransactionIsDestOSPEnabled (result->handle, &ospenabled) && (ospenabled == OSPE_OSP_FALSE)) {
- result->token[0] = 0;
- }
- else {
- ast_base64encode(result->token, token, tokenlen, sizeof(result->token) - 1);
- }
- if ((strlen(destination) > 2) && !OSPPTransactionGetDestProtocol(result->handle, &prot)) {
- res = 1;
+ sizeof(callednum), callednum, sizeof(callingnum), callingnum, sizeof(destination), destination, 0, NULL, &tokenlen, token))
+ {
+ if ((strlen(destination) > 2) && check_dest(result, token, tokenlen)) {
/* Strip leading and trailing brackets */
destination[strlen(destination) - 1] = '\0';
- switch(prot) {
- case OSPE_DEST_PROT_H323_SETUP:
- ast_copy_string(result->tech, "H323", sizeof(result->tech));
- snprintf(result->dest, sizeof(result->dest), "%s@%s", callednum, destination + 1);
- break;
- case OSPE_DEST_PROT_SIP:
- ast_copy_string(result->tech, "SIP", sizeof(result->tech));
- snprintf(result->dest, sizeof(result->dest), "%s@%s", callednum, destination + 1);
- break;
- case OSPE_DEST_PROT_IAX:
- ast_copy_string(result->tech, "IAX", sizeof(result->tech));
- snprintf(result->dest, sizeof(result->dest), "%s@%s", callednum, destination + 1);
- break;
- default:
- ast_log(LOG_DEBUG, "Unknown destination protocol '%d', skipping...\n", prot);
- res = 0;
- }
- } else {
- ast_log(LOG_DEBUG, "Missing destination protocol\n");
- break;
+ snprintf(result->dest, sizeof(result->dest), "%s@%s", callednum, destination + 1);
+ res = 1;
}
}
}
-
}
if (!res) {
OSPPTransactionDelete(result->handle);
result->handle = -1;
}
-
}
return res;
}
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