[svn-commits] branch oej/siptransfer r27721 - in
/team/oej/siptransfer: ./ apps/ channels/ ...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Wed May 17 00:33:33 MST 2006
Author: oej
Date: Wed May 17 02:33:33 2006
New Revision: 27721
URL: http://svn.digium.com/view/asterisk?rev=27721&view=rev
Log:
Updating to trunk
Modified:
team/oej/siptransfer/ (props changed)
team/oej/siptransfer/apps/app_channelredirect.c
team/oej/siptransfer/apps/app_dial.c
team/oej/siptransfer/apps/app_settransfercapability.c
team/oej/siptransfer/apps/app_voicemail.c
team/oej/siptransfer/channel.c
team/oej/siptransfer/channels/chan_misdn.c
team/oej/siptransfer/channels/chan_sip.c
team/oej/siptransfer/channels/chan_zap.c
team/oej/siptransfer/channels/misdn/isdn_lib.c
team/oej/siptransfer/channels/misdn/isdn_lib.h
team/oej/siptransfer/channels/misdn/isdn_msg_parser.c
team/oej/siptransfer/funcs/func_odbc.c
team/oej/siptransfer/include/asterisk/frame.h
team/oej/siptransfer/include/asterisk/rtp.h
team/oej/siptransfer/rtp.c
Propchange: team/oej/siptransfer/
------------------------------------------------------------------------------
Binary property 'branch-1.2-blocked' - no diff available.
Propchange: team/oej/siptransfer/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.
Propchange: team/oej/siptransfer/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Wed May 17 02:33:33 2006
@@ -1,1 +1,1 @@
-/trunk:1-27339
+/trunk:1-27720
Modified: team/oej/siptransfer/apps/app_channelredirect.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/apps/app_channelredirect.c?rev=27721&r1=27720&r2=27721&view=diff
==============================================================================
--- team/oej/siptransfer/apps/app_channelredirect.c (original)
+++ team/oej/siptransfer/apps/app_channelredirect.c Wed May 17 02:33:33 2006
@@ -40,6 +40,7 @@
#include "asterisk/lock.h"
#include "asterisk/app.h"
#include "asterisk/features.h"
+#include "asterisk/options.h"
static char *app = "ChannelRedirect";
static char *synopsis = "Redirects given channel to a dialplan target.";
@@ -105,7 +106,8 @@
goto chanquit;
}
- ast_log(LOG_DEBUG, "Attempting async goto (%s) to %s|%s|%d\n", args.channel, S_OR(context, chan2->context), S_OR(exten, chan2->exten), prio);
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "Attempting async goto (%s) to %s|%s|%d\n", args.channel, S_OR(context, chan2->context), S_OR(exten, chan2->exten), prio);
if (ast_async_goto_if_exists(chan2, S_OR(context, chan2->context), S_OR(exten, chan2->exten), prio))
ast_log(LOG_WARNING, "%s failed for %s\n", app, args.channel);
Modified: team/oej/siptransfer/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/apps/app_dial.c?rev=27721&r1=27720&r2=27721&view=diff
==============================================================================
--- team/oej/siptransfer/apps/app_dial.c (original)
+++ team/oej/siptransfer/apps/app_dial.c Wed May 17 02:33:33 2006
@@ -472,6 +472,8 @@
c = o->chan = ast_request(tech, in->nativeformats, stuff, &cause);
if (!c)
ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s/%s' (cause = %d)\n", tech, stuff, cause);
+ else
+ ast_channel_inherit_variables(in, o->chan);
} else {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Too many forwards from %s\n", c->name);
@@ -1051,6 +1053,8 @@
tmp->chan = ast_request(tech, chan->nativeformats, stuff, &cause);
if (!tmp->chan)
ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s/%s' (cause = %d)\n", tech, stuff, cause);
+ else
+ ast_channel_inherit_variables(chan, tmp->chan);
} else {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Too many forwards from %s\n", tmp->chan->name);
Modified: team/oej/siptransfer/apps/app_settransfercapability.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/apps/app_settransfercapability.c?rev=27721&r1=27720&r2=27721&view=diff
==============================================================================
--- team/oej/siptransfer/apps/app_settransfercapability.c (original)
+++ team/oej/siptransfer/apps/app_settransfercapability.c Wed May 17 02:33:33 2006
@@ -65,7 +65,7 @@
" RESTRICTED_DIGITAL : 0x09 - Restricted digital information\n"
" 3K1AUDIO : 0x10 - 3.1kHz Audio (fax calls)\n"
" DIGITAL_W_TONES : 0x11 - Unrestricted digital information with tones/announcements\n"
-" VIDEO : 0x18 - Video:\n"
+" VIDEO : 0x18 - Video\n"
"\n"
;
Modified: team/oej/siptransfer/apps/app_voicemail.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/apps/app_voicemail.c?rev=27721&r1=27720&r2=27721&view=diff
==============================================================================
--- team/oej/siptransfer/apps/app_voicemail.c (original)
+++ team/oej/siptransfer/apps/app_voicemail.c Wed May 17 02:33:33 2006
@@ -2413,17 +2413,17 @@
static int leave_voicemail(struct ast_channel *chan, char *ext, struct leave_vm_options *options)
{
- char tmptxtfile[256], txtfile[256];
+ char txtfile[256], tmptxtfile[256];
char callerid[256];
FILE *txt;
- int res = 0;
+ int res = 0, txtdes;
int msgnum;
int duration = 0;
int ausemacro = 0;
int ousemacro = 0;
int ouseexten = 0;
char date[256];
- char dir[256];
+ char dir[256], tmpdir[260];
char fn[256];
char prefile[256]="";
char tempfile[256]="";
@@ -2475,6 +2475,7 @@
DISPOSE(tempfile, -1);
/* It's easier just to try to make it than to check for its existence */
create_dirpath(dir, sizeof(dir), vmu->context, ext, "INBOX");
+ create_dirpath(tmpdir, sizeof(tmpdir), vmu->context, ext, "tmp");
/* Check current or macro-calling context for special extensions */
if (ast_test_flag(vmu, VM_OPERATOR)) {
@@ -2579,121 +2580,129 @@
if (!ast_strlen_zero(fmt)) {
msgnum = 0;
- if (vm_lock_path(dir)) {
- free_user(vmu);
- return ERROR_LOCK_PATH;
- }
-
- /*
- * This operation can be very expensive if done say over NFS or if the mailbox has 100+ messages
- * in the folder. So we should get this first so we don't cut off the first few seconds of the
- * message.
- */
- do {
- make_file(fn, sizeof(fn), dir, msgnum);
- if (!EXISTS(dir,msgnum,fn,chan->language))
- break;
- msgnum++;
- } while (msgnum < vmu->maxmsg);
+ if (count_messages(vmu, dir) >= vmu->maxmsg) {
+ res = ast_streamfile(chan, "vm-mailboxfull", chan->language);
+ if (!res)
+ res = ast_waitstream(chan, "");
+ ast_log(LOG_WARNING, "No more messages possible\n");
+ pbx_builtin_setvar_helper(chan, "VMSTATUS", "FAILED");
+ goto leave_vm_out;
+ }
+
+ snprintf(tmptxtfile, sizeof(tmptxtfile), "%s/XXXXXX", tmpdir);
+ txtdes = mkstemp(tmptxtfile);
+ if (txtdes < 0) {
+ res = ast_streamfile(chan, "vm-mailboxfull", chan->language);
+ if (!res)
+ res = ast_waitstream(chan, "");
+ ast_log(LOG_ERROR, "Unable to create message file: %s\n", strerror(errno));
+ pbx_builtin_setvar_helper(chan, "VMSTATUS", "FAILED");
+ goto leave_vm_out;
+ }
/* Now play the beep once we have the message number for our next message. */
if (res >= 0) {
/* Unless we're *really* silent, try to send the beep */
res = ast_stream_and_wait(chan, "beep", chan->language, "");
}
- if (msgnum < vmu->maxmsg) {
- /* assign a variable with the name of the voicemail file */
- pbx_builtin_setvar_helper(chan, "VM_MESSAGEFILE", fn);
-
- /* Store information */
- snprintf(txtfile, sizeof(txtfile), "%s.txt", fn);
- snprintf(tmptxtfile, sizeof(tmptxtfile), "%s.txt.tmp", fn);
- txt = fopen(tmptxtfile, "w+");
- if (txt) {
- get_date(date, sizeof(date));
- fprintf(txt,
- ";\n"
- "; Message Information file\n"
- ";\n"
- "[message]\n"
- "origmailbox=%s\n"
- "context=%s\n"
- "macrocontext=%s\n"
- "exten=%s\n"
- "priority=%d\n"
- "callerchan=%s\n"
- "callerid=%s\n"
- "origdate=%s\n"
- "origtime=%ld\n"
- "category=%s\n",
- ext,
- chan->context,
- chan->macrocontext,
- chan->exten,
- chan->priority,
- chan->name,
- ast_callerid_merge(callerid, sizeof(callerid), chan->cid.cid_name, chan->cid.cid_num, "Unknown"),
- date, (long)time(NULL),
- category ? category : "");
- } else
- ast_log(LOG_WARNING, "Error opening text file for output\n");
- res = play_record_review(chan, NULL, fn, vmmaxmessage, fmt, 1, vmu, &duration, dir, options->record_gain);
- if (res == '0') {
- if (txt && EXISTS(dir,msgnum,fn,chan->language)) {
- fclose(txt);
- rename(tmptxtfile, txtfile);
- } else if (txt && !EXISTS(dir,msgnum,fn,chan->language)) {
- if (option_debug)
- ast_log(LOG_DEBUG, "The recorded media file is gone, so we should remove the .txt file too!\n");
- fclose(txt);
- unlink(tmptxtfile);
- }
- goto transfer;
- }
- if (res > 0)
- res = 0;
- if (txt) {
- fprintf(txt, "duration=%d\n", duration);
- fclose(txt);
- rename(tmptxtfile, txtfile);
- }
-
+
+ /* Store information */
+ txt = fdopen(txtdes, "w+");
+ if (txt) {
+ get_date(date, sizeof(date));
+ fprintf(txt,
+ ";\n"
+ "; Message Information file\n"
+ ";\n"
+ "[message]\n"
+ "origmailbox=%s\n"
+ "context=%s\n"
+ "macrocontext=%s\n"
+ "exten=%s\n"
+ "priority=%d\n"
+ "callerchan=%s\n"
+ "callerid=%s\n"
+ "origdate=%s\n"
+ "origtime=%ld\n"
+ "category=%s\n",
+ ext,
+ chan->context,
+ chan->macrocontext,
+ chan->exten,
+ chan->priority,
+ chan->name,
+ ast_callerid_merge(callerid, sizeof(callerid), chan->cid.cid_name, chan->cid.cid_num, "Unknown"),
+ date, (long)time(NULL),
+ category ? category : "");
+ } else
+ ast_log(LOG_WARNING, "Error opening text file for output\n");
+ res = play_record_review(chan, NULL, tmptxtfile, vmmaxmessage, fmt, 1, vmu, &duration, NULL, options->record_gain);
+
+ if (txt) {
if (duration < vmminmessage) {
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "Recording was %d seconds long but needs to be at least %d - abandoning\n", duration, vmminmessage);
DELETE(dir,msgnum,fn);
- /* XXX We should really give a prompt too short/option start again, with leave_vm_out called only after a timeout XXX */
- pbx_builtin_setvar_helper(chan, "VMSTATUS", "FAILED");
- goto leave_vm_out;
- }
- /* Are there to be more recipients of this message? */
- while (tmpptr) {
- struct ast_vm_user recipu, *recip;
- char *exten, *context;
-
- exten = strsep(&tmpptr, "&");
- context = strchr(exten, '@');
- if (context) {
- *context = '\0';
- context++;
+ } else {
+ fprintf(txt, "duration=%d\n", duration);
+ fclose(txt);
+ if (vm_lock_path(dir)) {
+ ast_log(LOG_ERROR, "Couldn't lock directory %s. Voicemail will be lost.\n", dir);
+ /* Delete files */
+ ast_filedelete(tmptxtfile, NULL);
+ unlink(tmptxtfile);
+ } else {
+ for (;;) {
+ make_file(fn, sizeof(fn), dir, msgnum);
+ if (!EXISTS(dir, msgnum, fn, NULL))
+ break;
+ msgnum++;
+ }
+
+ /* assign a variable with the name of the voicemail file */
+ pbx_builtin_setvar_helper(chan, "VM_MESSAGEFILE", fn);
+
+ snprintf(txtfile, sizeof(txtfile), "%s.txt", fn);
+ ast_filerename(tmptxtfile, fn, NULL);
+ rename(tmptxtfile, txtfile);
+
+ ast_unlock_path(dir);
+
+ /* Are there to be more recipients of this message? */
+ while (tmpptr) {
+ struct ast_vm_user recipu, *recip;
+ char *exten, *context;
+
+ exten = strsep(&tmpptr, "&");
+ context = strchr(exten, '@');
+ if (context) {
+ *context = '\0';
+ context++;
+ }
+ if ((recip = find_user(&recipu, context, exten))) {
+ copy_message(chan, vmu, 0, msgnum, duration, recip, fmt);
+ free_user(recip);
+ }
+ }
+ if (ast_fileexists(fn, NULL, NULL)) {
+ STORE(dir, vmu->mailbox, vmu->context, msgnum);
+ notify_new_message(chan, vmu, msgnum, duration, fmt, chan->cid.cid_num, chan->cid.cid_name);
+ DISPOSE(dir, msgnum);
+ }
}
- if ((recip = find_user(&recipu, context, exten))) {
- copy_message(chan, vmu, 0, msgnum, duration, recip, fmt);
- free_user(recip);
- }
- }
- if (ast_fileexists(fn, NULL, NULL)) {
- STORE(dir, vmu->mailbox, vmu->context, msgnum);
- notify_new_message(chan, vmu, msgnum, duration, fmt, chan->cid.cid_num, chan->cid.cid_name);
- DISPOSE(dir, msgnum);
- }
+ }
+ }
+
+ if (res == '0') {
+ goto transfer;
+ } else if (res > 0)
+ res = 0;
+
+ if (duration < vmminmessage)
+ /* XXX We should really give a prompt too short/option start again, with leave_vm_out called only after a timeout XXX */
+ pbx_builtin_setvar_helper(chan, "VMSTATUS", "FAILED");
+ else
pbx_builtin_setvar_helper(chan, "VMSTATUS", "SUCCESS");
- } else {
- ast_unlock_path(dir);
- res = ast_stream_and_wait(chan, "vm-mailboxfull", chan->language, "");
- ast_log(LOG_WARNING, "No more messages possible\n");
- pbx_builtin_setvar_helper(chan, "VMSTATUS", "FAILED");
- }
} else
ast_log(LOG_WARNING, "No format for saving voicemail?\n");
leave_vm_out:
Modified: team/oej/siptransfer/channel.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/channel.c?rev=27721&r1=27720&r2=27721&view=diff
==============================================================================
--- team/oej/siptransfer/channel.c (original)
+++ team/oej/siptransfer/channel.c Wed May 17 02:33:33 2006
@@ -1956,9 +1956,11 @@
if (f->subclass == AST_CONTROL_ANSWER) {
if (!ast_test_flag(chan, AST_FLAG_OUTGOING)) {
ast_log(LOG_DEBUG, "Ignoring answer on an inbound call!\n");
+ ast_frfree(f);
f = &ast_null_frame;
} else if (prestate == AST_STATE_UP) {
ast_log(LOG_DEBUG, "Dropping duplicate answer!\n");
+ ast_frfree(f);
f = &ast_null_frame;
} else {
/* Answer the CDR */
@@ -1974,6 +1976,7 @@
chan->dtmfq[strlen(chan->dtmfq)] = f->subclass;
else
ast_log(LOG_WARNING, "Dropping deferred DTMF digits on %s\n", chan->name);
+ ast_frfree(f);
f = &ast_null_frame;
}
break;
Modified: team/oej/siptransfer/channels/chan_misdn.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/channels/chan_misdn.c?rev=27721&r1=27720&r2=27721&view=diff
==============================================================================
--- team/oej/siptransfer/channels/chan_misdn.c (original)
+++ team/oej/siptransfer/channels/chan_misdn.c Wed May 17 02:33:33 2006
@@ -105,6 +105,7 @@
+
/* BEGIN: chan_misdn.h */
@@ -206,6 +207,12 @@
};
static struct robin_list *robin = NULL;
+
+
+struct ast_frame *process_ast_dsp(struct chan_list *tmp, struct ast_frame *frame);
+
+
+
static inline void free_robin_list_r (struct robin_list *r)
{
if (r) {
@@ -300,6 +307,9 @@
static int misdn_facility_exec(struct ast_channel *chan, void *data);
int chan_misdn_jb_empty(struct misdn_bchannel *bc, char *buf, int len);
+
+
+void debug_numplan(int port, int numplan, char *type);
/*************** Helpers *****************/
@@ -1227,6 +1237,28 @@
}
+void debug_numplan(int port, int numplan, char *type)
+{
+ switch (numplan) {
+ case NUMPLAN_INTERNATIONAL:
+ chan_misdn_log(2, port, " --> %s: International\n",type);
+ break;
+ case NUMPLAN_NATIONAL:
+ chan_misdn_log(2, port, " --> %s: National\n",type);
+ break;
+ case NUMPLAN_SUBSCRIBER:
+ chan_misdn_log(2, port, " --> %s: Subscriber\n",type);
+ break;
+ case NUMPLAN_UNKNOWN:
+ chan_misdn_log(2, port, " --> %s: Unknown\n",type);
+ break;
+ /* Maybe we should cut off the prefix if present ? */
+ default:
+ chan_misdn_log(0, port, " --> !!!! Wrong dialplan setting, please see the misdn.conf sample file\n ");
+ break;
+ }
+}
+
static int read_config(struct chan_list *ch, int orig) {
if (!ch) {
@@ -1291,8 +1323,7 @@
misdn_cfg_get( bc->port, MISDN_CFG_CONTEXT, ch->context, sizeof(ch->context));
- ast_copy_string (ast->context,ch->context,sizeof(ast->context));
-
+ ast_copy_string (ast->context,ch->context,sizeof(ast->context));
{
int ec, ectr;
@@ -1311,7 +1342,6 @@
}
}
-
{
int eb3;
@@ -1321,7 +1351,6 @@
port=bc->port;
-
{
char buf[256];
ast_group_t pg,cg;
@@ -1334,10 +1363,8 @@
ast->callgroup=cg;
}
-
if ( orig == ORG_AST) {
misdn_cfg_get( port, MISDN_CFG_TE_CHOOSE_CHANNEL, &(bc->te_choose_channel), sizeof(int));
-
{
char callerid[BUFFERSIZE+1];
@@ -1354,67 +1381,11 @@
misdn_cfg_get( port, MISDN_CFG_DIALPLAN, &bc->dnumplan, sizeof(int));
- switch (bc->dnumplan) {
- case NUMPLAN_INTERNATIONAL:
- chan_misdn_log(2, port, " --> TON: International\n");
- break;
- case NUMPLAN_NATIONAL:
- chan_misdn_log(2, port, " --> TON: National\n");
- break;
- case NUMPLAN_SUBSCRIBER:
- chan_misdn_log(2, port, " --> TON: Subscriber\n");
- break;
- case NUMPLAN_UNKNOWN:
- chan_misdn_log(2, port, " --> TON: Unknown\n");
- break;
- /* Maybe we should cut off the prefix if present ? */
- default:
- chan_misdn_log(0, port, " --> !!!! Wrong dialplan setting, please see the misdn.conf sample file\n ");
- break;
- }
-
-
misdn_cfg_get( port, MISDN_CFG_LOCALDIALPLAN, &bc->onumplan, sizeof(int));
- switch (bc->onumplan) {
- case NUMPLAN_INTERNATIONAL:
- chan_misdn_log(2, port, " --> LTON: International\n");
- break;
- case NUMPLAN_NATIONAL:
- chan_misdn_log(2, port, " --> LTON: National\n");
- break;
- case NUMPLAN_SUBSCRIBER:
- chan_misdn_log(2, port, " --> LTON: Subscriber\n");
- break;
- case NUMPLAN_UNKNOWN:
- chan_misdn_log(2, port, " --> LTON: Unknown\n");
- break;
- /* Maybe we should cut off the prefix if present ? */
- default:
- chan_misdn_log(0, port, " --> !!!! Wrong dialplan setting, please see the misdn.conf sample file\n ");
- break;
- }
-
misdn_cfg_get( port, MISDN_CFG_CPNDIALPLAN, &bc->cpnnumplan, sizeof(int));
-
- switch (bc->cpnnumplan) {
- case NUMPLAN_INTERNATIONAL:
- chan_misdn_log(2, port, " --> CTON: International\n");
- break;
- case NUMPLAN_NATIONAL:
- chan_misdn_log(2, port, " --> CTON: National\n");
- break;
- case NUMPLAN_SUBSCRIBER:
- chan_misdn_log(2, port, " --> CTON: Subscriber\n");
- break;
- case NUMPLAN_UNKNOWN:
- chan_misdn_log(2, port, " --> CTON: Unknown\n");
- break;
- /* Maybe we should cut off the prefix if present ? */
- default:
- chan_misdn_log(0, port, " --> !!!! Wrong dialplan setting, please see the misdn.conf sample file\n ");
- break;
- }
-
+ debug_numplan(port, bc->dnumplan,"TON");
+ debug_numplan(port, bc->onumplan,"LTON");
+ debug_numplan(port, bc->cpnnumplan,"CTON");
}
@@ -1422,25 +1393,7 @@
} else { /** ORIGINATOR MISDN **/
misdn_cfg_get( port, MISDN_CFG_CPNDIALPLAN, &bc->cpnnumplan, sizeof(int));
-
- switch (bc->cpnnumplan) {
- case NUMPLAN_INTERNATIONAL:
- chan_misdn_log(2, port, " --> CTON: International\n");
- break;
- case NUMPLAN_NATIONAL:
- chan_misdn_log(2, port, " --> CTON: National\n");
- break;
- case NUMPLAN_SUBSCRIBER:
- chan_misdn_log(2, port, " --> CTON: Subscriber\n");
- break;
- case NUMPLAN_UNKNOWN:
- chan_misdn_log(2, port, " --> CTON: Unknown\n");
- break;
- /* Maybe we should cut off the prefix if present ? */
- default:
- chan_misdn_log(0, port, " --> !!!! Wrong dialplan setting, please see the misdn.conf sample file\n ");
- break;
- }
+ debug_numplan(port, bc->cpnnumplan,"CTON");
char prefix[BUFFERSIZE+1]="";
switch( bc->onumplan ) {
@@ -1451,14 +1404,6 @@
case NUMPLAN_NATIONAL:
misdn_cfg_get( bc->port, MISDN_CFG_NATPREFIX, prefix, BUFFERSIZE);
break;
-
-
- case NUMPLAN_SUBSCRIBER:
- /* dunno what to do here ? */
- break;
-
- case NUMPLAN_UNKNOWN:
- break;
default:
break;
}
@@ -1471,7 +1416,6 @@
strcpy(bc->oad,tmp);
}
-
if (!ast_strlen_zero(bc->dad)) {
ast_copy_string(bc->orig_dad,bc->dad, sizeof(bc->orig_dad));
}
@@ -1479,23 +1423,15 @@
if ( ast_strlen_zero(bc->dad) && !ast_strlen_zero(bc->keypad)) {
ast_copy_string(bc->dad,bc->keypad, sizeof(bc->dad));
}
+
prefix[0] = 0;
switch( bc->dnumplan ) {
case NUMPLAN_INTERNATIONAL:
misdn_cfg_get( bc->port, MISDN_CFG_INTERNATPREFIX, prefix, BUFFERSIZE);
break;
-
- case NUMPLAN_NATIONAL:
- misdn_cfg_get( bc->port, MISDN_CFG_NATPREFIX, prefix, BUFFERSIZE);
- break;
-
-
- case NUMPLAN_SUBSCRIBER:
- /* dunno what to do here ? */
- break;
-
- case NUMPLAN_UNKNOWN:
+ case NUMPLAN_NATIONAL:
+ misdn_cfg_get( bc->port, MISDN_CFG_NATPREFIX, prefix, BUFFERSIZE);
break;
default:
break;
@@ -1520,7 +1456,8 @@
free(ast->cid.cid_rdnis);
ast->cid.cid_rdnis = strdup(bc->rad);
}
- }
+ } /* ORIG MISDN END */
+
return 0;
}
@@ -1552,10 +1489,7 @@
chan_misdn_log(-1,0,"misdn_call: No Extension given!\n");
return -1;
}
-
- }
-
-
+ }
}
if (!ast) {
@@ -1570,7 +1504,6 @@
return -1;
}
-
if (!ch) {
ast_log(LOG_WARNING, " --> ! misdn_call called on %s, neither down nor reserved (or dest==NULL)\n", ast->name);
ast->hangupcause=41;
@@ -1591,12 +1524,9 @@
strncpy(newbc->dad,ext,sizeof( newbc->dad));
strncpy(ast->exten,ext,sizeof(ast->exten));
-
chan_misdn_log(1, port, "* CALL: %s\n",dest);
chan_misdn_log(1, port, " --> * dad:%s tech:%s ctx:%s\n",ast->exten,ast->name, ast->context);
-
-
chan_misdn_log(3, port, " --> * adding2newbc ext %s\n",ast->exten);
if (ast->exten) {
@@ -1637,7 +1567,6 @@
misdn_set_opt_exec(ast,opts);
else
chan_misdn_log(2,port,"NO OPTS GIVEN\n");
-
ch->state=MISDN_CALLING;
@@ -1798,7 +1727,7 @@
-static int misdn_indication(struct ast_channel *ast, int cond)
+static int misdn_indication(struct ast_channel *ast, int cond, const void *data, size_t datalen)
{
struct chan_list *p;
@@ -2051,52 +1980,69 @@
return 0;
}
+
+struct ast_frame *process_ast_dsp(struct chan_list *tmp, struct ast_frame *frame)
+{
+ struct ast_frame *f,*f2;
+ if (tmp->trans)
+ f2=ast_translate(tmp->trans, frame,0);
+ else {
+ chan_misdn_log(0, tmp->bc->port, "No T-Path found\n");
+ return NULL;
+ }
+
+ f = ast_dsp_process(tmp->ast, tmp->dsp, f2);
+ if (f && (f->frametype == AST_FRAME_DTMF)) {
+ ast_log(LOG_DEBUG, "Detected inband DTMF digit: %c", f->subclass);
+ if (f->subclass == 'f' && tmp->faxdetect) {
+ /* Fax tone -- Handle and return NULL */
+ struct ast_channel *ast = tmp->ast;
+ if (!tmp->faxhandled) {
+ tmp->faxhandled++;
+ if (strcmp(ast->exten, "fax")) {
+ if (ast_exists_extension(ast, ast_strlen_zero(ast->macrocontext)? ast->context : ast->macrocontext, "fax", 1, AST_CID_P(ast))) {
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_3 "Redirecting %s to fax extension\n", ast->name);
+ /* Save the DID/DNIS when we transfer the fax call to a "fax" extension */
+ pbx_builtin_setvar_helper(ast,"FAXEXTEN",ast->exten);
+ if (ast_async_goto(ast, ast->context, "fax", 1))
+ ast_log(LOG_WARNING, "Failed to async goto '%s' into fax of '%s'\n", ast->name, ast->context);
+ } else
+ ast_log(LOG_NOTICE, "Fax detected, but no fax extension ctx:%s exten:%s\n",ast->context, ast->exten);
+ } else
+ ast_log(LOG_DEBUG, "Already in a fax extension, not redirecting\n");
+ } else
+ ast_log(LOG_DEBUG, "Fax already handled\n");
+
+ } else if ( tmp->ast_dsp) {
+ chan_misdn_log(2, tmp->bc->port, " --> * SEND: DTMF (AST_DSP) :%c\n",f->subclass);
+ return f;
+ }
+ }
+
+ frame->frametype = AST_FRAME_NULL;
+ frame->subclass = 0;
+ return frame;
+}
+
+
static struct ast_frame *misdn_read(struct ast_channel *ast)
{
struct chan_list *tmp;
-
- char blah[255];
- int len =0 ;
+ int len;
if (!ast) return NULL;
if (! (tmp=MISDN_ASTERISK_TECH_PVT(ast)) ) return NULL;
if (!tmp->bc) return NULL;
-
- read(tmp->pipe[0],blah,sizeof(blah));
-
- len = misdn_ibuf_usedcount(tmp->bc->astbuf);
-
- if (!len) {
- struct ast_frame *frame;
- if(!tmp->zero_read_cnt)
- chan_misdn_log(4,tmp->bc->port,"misdn_read: ZERO READ\n");
- tmp->zero_read_cnt++;
-
- if (tmp->zero_read_cnt > 5000) {
- chan_misdn_log(4,tmp->bc->port,"misdn_read: ZERO READ counted > 5000 times\n");
- tmp->zero_read_cnt=0;
-
- }
-
- /*faking Voice Frame*/
- tmp->frame.frametype = AST_FRAME_VOICE;
- tmp->frame.subclass = AST_FORMAT_ALAW;
- memset(tmp->ast_rd_buf,0,128);
- tmp->frame.data = tmp->ast_rd_buf ;
- tmp->frame.mallocd =0 ;
- tmp->frame.datalen = 128;
- tmp->frame.samples = 128;
-
- frame=ast_frisolate(&tmp->frame);
- return frame;
- }
-
- /*shrinken len if necessary, we transmit at maximum 4k*/
- len = len<=sizeof(tmp->ast_rd_buf)?len:sizeof(tmp->ast_rd_buf);
-
- misdn_ibuf_memcpy_r(tmp->ast_rd_buf, tmp->bc->astbuf,len);
-
+ len=read(tmp->pipe[0],tmp->ast_rd_buf,sizeof(tmp->ast_rd_buf));
+
+ if (len<=0) {
+ /* we hangup here, since our pipe is closed */
+ chan_misdn_log(2,tmp->bc->port,"misdn_read: Pipe closed, hanging up\n");
+ return NULL;
+ }
+
tmp->frame.frametype = AST_FRAME_VOICE;
tmp->frame.subclass = AST_FORMAT_ALAW;
tmp->frame.datalen = len;
@@ -2105,11 +2051,13 @@
tmp->frame.offset= 0 ;
tmp->frame.src = NULL;
tmp->frame.data = tmp->ast_rd_buf ;
-
+
+ if (tmp->faxdetect || tmp->ast_dsp ) {
+ return process_ast_dsp(tmp, &tmp->frame);
+ }
+
return &tmp->frame;
}
-
-
static int misdn_write(struct ast_channel *ast, struct ast_frame *frame)
@@ -2123,10 +2071,6 @@
ast_log(LOG_WARNING, "private but no bc\n");
return -1;
}
-
- /*if (ch->bc->tone != TONE_NONE)
- tone_indicate(ch,TONE_NONE); */
-
if (ch->holded ) {
chan_misdn_log(5, ch->bc->port, "misdn_write: Returning because holded\n");
@@ -2192,10 +2136,6 @@
}
chan_misdn_log(9, ch->bc->port, "Sending :%d bytes 2 MISDN\n",frame->samples);
- /*if speech flip bits*/
- if ( misdn_cap_is_speech(ch->bc->capability) )
- flip_buf_bits(frame->data,frame->samples);
-
if ( !ch->bc->nojitter && misdn_cap_is_speech(ch->bc->capability) ) {
/* Buffered Transmit (triggert by read from isdn side)*/
@@ -2676,116 +2616,6 @@
return tmp;
}
-
-
-
-static int misdn_tx2ast_frm(struct chan_list * tmp, char * buf, int len )
-{
- struct ast_frame frame;
-
- /* If in hold state we drop frame .. */
- if (tmp->holded ) return 0;
-
- switch(tmp->state) {
- case MISDN_CLEANING:
- case MISDN_EXTCANTMATCH:
- return 0;
-
- case MISDN_WAITING4DIGS:
- default:
- break;
- }
-
- if (tmp->norxtone) {
- chan_misdn_log(3, tmp->bc->port, "misdn_tx2ast_frm: Returning because norxtone\n");
- return 0;
- }
-
- frame.frametype = AST_FRAME_VOICE;
- frame.subclass = AST_FORMAT_ALAW;
- frame.datalen = len;
- frame.samples = len ;
- frame.mallocd =0 ;
- frame.offset= 0 ;
- frame.src = NULL;
- frame.data = buf ;
-
- if (tmp->faxdetect || tmp->ast_dsp ) {
- struct ast_frame *f,*f2;
- if (tmp->trans)
- f2=ast_translate(tmp->trans, &frame,0);
- else {
- chan_misdn_log(0, tmp->bc->port, "No T-Path found\n");
- return 0;
- }
-
- f = ast_dsp_process(tmp->ast, tmp->dsp, f2);
- if (f && (f->frametype == AST_FRAME_DTMF)) {
- ast_log(LOG_DEBUG, "Detected inband DTMF digit: %c", f->subclass);
- if (f->subclass == 'f' && tmp->faxdetect) {
- /* Fax tone -- Handle and return NULL */
- struct ast_channel *ast = tmp->ast;
- if (!tmp->faxhandled) {
- tmp->faxhandled++;
- if (strcmp(ast->exten, "fax")) {
- if (ast_exists_extension(ast, S_OR(ast->macrocontext, ast->context), "fax", 1, AST_CID_P(ast))) {
- if (option_verbose > 2)
- ast_verbose(VERBOSE_PREFIX_3 "Redirecting %s to fax extension\n", ast->name);
- /* Save the DID/DNIS when we transfer the fax call to a "fax" extension */
- pbx_builtin_setvar_helper(ast,"FAXEXTEN",ast->exten);
- if (ast_async_goto(ast, ast->context, "fax", 1))
- ast_log(LOG_WARNING, "Failed to async goto '%s' into fax of '%s'\n", ast->name, ast->context);
- } else
- ast_log(LOG_NOTICE, "Fax detected, but no fax extension ctx:%s exten:%s\n",ast->context, ast->exten);
- } else
- ast_log(LOG_DEBUG, "Already in a fax extension, not redirecting\n");
- } else
- ast_log(LOG_DEBUG, "Fax already handled\n");
- frame.frametype = AST_FRAME_NULL;
- frame.subclass = 0;
- f = &frame;
- } else if ( tmp->ast_dsp) {
- struct ast_frame fr;
- memset(&fr, 0 , sizeof(fr));
- fr.frametype = AST_FRAME_DTMF;
- fr.subclass = f->subclass ;
- fr.src=NULL;
- fr.data = NULL ;
- fr.datalen = 0;
- fr.samples = 0 ;
- fr.mallocd =0 ;
- fr.offset= 0 ;
-
- chan_misdn_log(2, tmp->bc->port, " --> * SEND: DTMF (AST_DSP) :%c\n",f->subclass);
- ast_queue_frame(tmp->ast, &fr);
-
- frame.frametype = AST_FRAME_NULL;
- frame.subclass = 0;
- f = &frame;
- }
- }
- }
-
- if (tmp && tmp->ast && MISDN_ASTERISK_PVT (tmp->ast) && MISDN_ASTERISK_TECH_PVT(tmp->ast) ) {
-#if MISDN_DEBUG
- int i, max=5>len?len:5;
-
- printf("write2* %p %d bytes: ",tmp, len);
-
- for (i=0; i< max ; i++) printf("%2.2x ",((char*) frame.data)[i]);
- printf ("\n");
-#endif
- chan_misdn_log(9, tmp->bc->port, "Queueing %d bytes 2 Asterisk\n",len);
- ast_queue_frame(tmp->ast,&frame);
-
- } else {
- ast_log (LOG_WARNING, "No ast || ast->pvt || ch\n");
- }
-
- return 0;
-}
-
-/** Channel Queue ***/
static struct chan_list *find_chan_by_l3id(struct chan_list *list, unsigned long l3id)
{
@@ -3697,40 +3527,24 @@
case EVENT_BCHAN_DATA:
{
- if ( !misdn_cap_is_speech(ch->bc->capability) || bc->nojitter) {
- misdn_tx2ast_frm(ch, bc->bframe, bc->bframe_len );
+ if ( !misdn_cap_is_speech(ch->bc->capability) ) {
+ struct ast_frame frame;
+ /*In Data Modes we queue frames*/
+ frame.frametype = AST_FRAME_VOICE; /*we have no data frames yet*/
+ frame.subclass = AST_FORMAT_ALAW;
+ frame.datalen = bc->bframe_len;
+ frame.samples = bc->bframe_len ;
+ frame.mallocd =0 ;
+ frame.offset= 0 ;
+ frame.src = NULL;
+ frame.data = bc->bframe ;
+
+ ast_queue_frame(ch->ast,&frame);
} else {
- int len=bc->bframe_len;
- int free=misdn_ibuf_freecount(bc->astbuf);
-
- if (bc->bframe_len > free) {
- ast_log(LOG_DEBUG, "sbuf overflow!\n");
- len=misdn_ibuf_freecount(bc->astbuf);
-
- if (len == 0) {
- ast_log(LOG_WARNING, "BCHAN_DATA: write buffer overflow port:%d channel:%d!\n",bc->port,bc->channel);
- }
- }
-
- misdn_ibuf_memcpy_w(bc->astbuf, bc->bframe, len);
-
- {
- char blah[1]="\0";
-#ifdef FLATTEN_JITTER
- {
- struct timeval tv;
- gettimeofday(&tv,NULL);
-
- if (tv.tv_usec % 10000 > 0 ) {
- write(ch->pipe[1], blah,sizeof(blah));
- bc->time_usec=tv.tv_usec;
- }
- }
-#else
- write(ch->pipe[1], blah,sizeof(blah));
-#endif
-
-
+ int ret=write(ch->pipe[1], bc->bframe, bc->bframe_len);
+
+ if (ret<=0) {
+ chan_misdn_log(1, bc->port, "Write returned <=0 (err=%s)\n",strerror(errno));
}
}
}
Modified: team/oej/siptransfer/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/channels/chan_sip.c?rev=27721&r1=27720&r2=27721&view=diff
==============================================================================
--- team/oej/siptransfer/channels/chan_sip.c (original)
+++ team/oej/siptransfer/channels/chan_sip.c Wed May 17 02:33:33 2006
@@ -2197,6 +2197,7 @@
ast_rtp_destroy(r->vrtp);
r->vrtp = NULL;
}
+ ast_rtp_setdtmf(r->rtp, ast_test_flag(&r->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
r->prefs = peer->prefs;
natflags = ast_test_flag(&r->flags[0], SIP_NAT) & SIP_NAT_ROUTE;
if (r->rtp) {
@@ -3572,9 +3573,12 @@
free(p);
return NULL;
}
+ ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
ast_rtp_settos(p->rtp, global_tos_audio);
- if (p->vrtp)
+ if (p->vrtp) {
ast_rtp_settos(p->vrtp, global_tos_video);
+ ast_rtp_setdtmf(p->vrtp, 0);
+ }
p->rtptimeout = global_rtptimeout;
p->rtpholdtimeout = global_rtpholdtimeout;
p->rtpkeepalive = global_rtpkeepalive;
@@ -4794,6 +4798,7 @@
char c[256];
char t[256];
char b[256];
+ char *hold;
char m_audio[256];
char m_video[256];
char a_audio[1024];
@@ -4873,6 +4878,11 @@
ast_build_string(&m_audio_next, &m_audio_left, "m=audio %d RTP/AVP", ntohs(dest.sin_port));
ast_build_string(&m_video_next, &m_video_left, "m=video %d RTP/AVP", ntohs(vdest.sin_port));
+ if (ast_test_flag(&p->flags[0], SIP_CALL_ONHOLD))
+ hold = "a=recvonly";
+ else
+ hold = "a=sendrecv";
+
/* Prefer the codec we were requested to use, first, no matter what */
if (capability & p->prefcodec) {
if (p->prefcodec <= AST_FORMAT_MAX_AUDIO)
@@ -4953,11 +4963,11 @@
ast_build_string(&m_audio_next, &m_audio_left, "\r\n");
ast_build_string(&m_video_next, &m_video_left, "\r\n");
- len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + strlen(m_audio) + strlen(a_audio);
+ len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + strlen(m_audio) + strlen(a_audio) + strlen(hold);
if ((p->vrtp) &&
(!ast_test_flag(&p->flags[0], SIP_NOVIDEO)) &&
(capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */
- len += strlen(m_video) + strlen(a_video) + strlen(b);
+ len += strlen(m_video) + strlen(a_video) + strlen(b) + strlen(hold);
add_header(resp, "Content-Type", "application/sdp");
add_header_contentLength(resp, len);
@@ -4972,11 +4982,13 @@
add_line(resp, t);
add_line(resp, m_audio);
add_line(resp, a_audio);
+ add_line(resp, hold);
if ((p->vrtp) &&
(!ast_test_flag(&p->flags[0], SIP_NOVIDEO)) &&
(capability & VIDEO_CODEC_MASK)) { /* only if video response is appropriate */
add_line(resp, m_video);
add_line(resp, a_video);
+ add_line(resp, hold);
}
/* Update lastrtprx when we send our SDP */
@@ -9132,6 +9144,7 @@
ast_mutex_lock(&iflock);
for (cur = iflist; cur; cur = cur->next) {
if (!strncasecmp(cur->callid, argv[3], len)) {
+ char formatbuf[BUFSIZ/2];
ast_cli(fd,"\n");
if (cur->subscribed != NONE)
ast_cli(fd, " * Subscription (type: %s)\n", subscription_type2str(cur->subscribed));
@@ -9145,10 +9158,10 @@
ast_cli(fd, " Owner channel ID: %s\n", "<none>");
}
ast_cli(fd, " Our Codec Capability: %d\n", cur->capability);
- ast_cli(fd, " Non-Codec Capability: %d\n", cur->noncodeccapability);
+ ast_cli(fd, " Non-Codec Capability (DTMF): %d\n", cur->noncodeccapability);
ast_cli(fd, " Their Codec Capability: %d\n", cur->peercapability);
ast_cli(fd, " Joint Codec Capability: %d\n", cur->jointcapability);
- ast_cli(fd, " Format %s\n", ast_getformatname(cur->owner ? cur->owner->nativeformats : 0) );
+ ast_cli(fd, " Format %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->owner ? cur->owner->nativeformats : 0) );
ast_cli(fd, " Theoretical Address: %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), cur->sa.sin_addr), ntohs(cur->sa.sin_port));
ast_cli(fd, " Received Address: %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), cur->recv.sin_addr), ntohs(cur->recv.sin_port));
ast_cli(fd, " SIP Transfer mode: %s\n", transfermode2str(cur->allowtransfer));
@@ -11533,6 +11546,7 @@
get_rdnis(p, NULL); /* Get redirect information */
extract_uri(p, req); /* Get the Contact URI */
build_contact(p); /* Build our contact header */
+ ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
if (!replace_id && gotdest) { /* No matching extension found */
if (gotdest == 1 && ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
Modified: team/oej/siptransfer/channels/chan_zap.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/channels/chan_zap.c?rev=27721&r1=27720&r2=27721&view=diff
==============================================================================
--- team/oej/siptransfer/channels/chan_zap.c (original)
+++ team/oej/siptransfer/channels/chan_zap.c Wed May 17 02:33:33 2006
@@ -2899,6 +2899,16 @@
}
ast_log(LOG_DEBUG, "Set Operator Services mode, value: %d on %s/%s\n",
oprmode->mode, chan->name,oprmode->peer->name);;
+ break;
+ case AST_OPTION_ECHOCAN:
+ cp = (char *) data;
+ if (*cp) {
+ ast_log(LOG_DEBUG, "Enabling echo cancelation on %s\n", chan->name);
+ zt_enable_ec(p);
+ } else {
+ ast_log(LOG_DEBUG, "Disabling echo cancelation on %s\n", chan->name);
+ zt_disable_ec(p);
+ }
break;
}
errno = 0;
Modified: team/oej/siptransfer/channels/misdn/isdn_lib.c
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/channels/misdn/isdn_lib.c?rev=27721&r1=27720&r2=27721&view=diff
==============================================================================
--- team/oej/siptransfer/channels/misdn/isdn_lib.c (original)
+++ team/oej/siptransfer/channels/misdn/isdn_lib.c Wed May 17 02:33:33 2006
@@ -19,42 +19,6 @@
void misdn_split_conf(struct misdn_bchannel *bc, int conf_id);
-void misdn_free_ibuffer(void *ibuf)
-{
- free_ibuffer((ibuffer_t*)ibuf);
-}
-
-
-void misdn_clear_ibuffer(void *ibuf)
-{
- clear_ibuffer( (ibuffer_t*)ibuf);
-}
-
-void *misdn_init_ibuffer(int len)
-{
- return init_ibuffer(len);
-}
-
-int misdn_ibuf_freecount(void *buf)
-{
- return ibuf_freecount( (ibuffer_t*)buf);
-}
-
-int misdn_ibuf_usedcount(void *buf)
-{
- return ibuf_usedcount( (ibuffer_t*)buf);
-}
-
-void misdn_ibuf_memcpy_r(char *to, void *buf, int len)
-{
- ibuf_memcpy_r( to, (ibuffer_t*)buf, len);
-}
-
-void misdn_ibuf_memcpy_w(void *buf, char *from, int len)
-{
- ibuf_memcpy_w((ibuffer_t*)buf, from, len);
-}
-
struct misdn_stack* get_misdn_stack( void );
@@ -196,7 +160,6 @@
void te_lib_destroy(int midev) ;
struct misdn_bchannel *manager_find_bc_by_pid(int pid);
struct misdn_bchannel *manager_find_bc_holded(struct misdn_bchannel* bc);
-unsigned char * manager_flip_buf_bits ( unsigned char * buf , int len);
void manager_ph_control_block(struct misdn_bchannel *bc, long c1, void *c2, int c2_len);
void manager_clean_bc(struct misdn_bchannel *bc );
void manager_bchannel_setup (struct misdn_bchannel *bc);
@@ -2031,6 +1994,7 @@
jlen=cb_jb_empty(bc,&buf[mISDN_HEADER_LEN],len);
if (jlen) {
+ flip_buf_bits( &buf[mISDN_HEADER_LEN], jlen);
if (jlen < len) {
cb_log(5,bc->port,"Jitterbuffer Underrun.\n");
@@ -3586,8 +3550,10 @@
frm->len = len;
memcpy(&buf[mISDN_HEADER_LEN], data,len);
-
- if ( ! misdn_cap_is_speech(bc->capability))
+
+ if ( misdn_cap_is_speech(bc->capability) )
+ flip_buf_bits( &buf[mISDN_HEADER_LEN], len);
+ else
cb_log(6, stack->port, "Writing %d data bytes\n",len);
cb_log(9, stack->port, "Writing %d bytes 2 mISDN\n",len);
Modified: team/oej/siptransfer/channels/misdn/isdn_lib.h
URL: http://svn.digium.com/view/asterisk/team/oej/siptransfer/channels/misdn/isdn_lib.h?rev=27721&r1=27720&r2=27721&view=diff
==============================================================================
--- team/oej/siptransfer/channels/misdn/isdn_lib.h (original)
+++ team/oej/siptransfer/channels/misdn/isdn_lib.h Wed May 17 02:33:33 2006
@@ -371,18 +371,6 @@
void get_show_stack_details(int port, char *buf);
-/** Ibuf interface **/
-int misdn_ibuf_usedcount(void *buf);
-int misdn_ibuf_freecount(void *buf);
-void misdn_ibuf_memcpy_r(char *to, void *from, int len);
-void misdn_ibuf_memcpy_w(void *buf, char *from, int len);
-
-void misdn_free_ibuffer(void *ibuf);
-void misdn_clear_ibuffer(void *ibuf);
-void *misdn_init_ibuffer(int len);
-
-/** Ibuf interface End **/
-
void misdn_lib_tone_generator_start(struct misdn_bchannel *bc);
[... 338 lines stripped ...]
More information about the svn-commits
mailing list