[svn-commits] oej: branch oej/codename-pineapple r48006 - in
/team/oej/codename-pineapple: ...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Sat Nov 25 05:03:47 MST 2006
Author: oej
Date: Sat Nov 25 06:03:46 2006
New Revision: 48006
URL: http://svn.digium.com/view/asterisk?view=rev&rev=48006
Log:
Update to trunk
Modified:
team/oej/codename-pineapple/ (props changed)
team/oej/codename-pineapple/CHANGES
team/oej/codename-pineapple/channels/chan_h323.c
team/oej/codename-pineapple/channels/chan_sip.c
team/oej/codename-pineapple/doc/asterisk-conf.txt
team/oej/codename-pineapple/include/asterisk/causes.h
team/oej/codename-pineapple/include/asterisk/channel.h
team/oej/codename-pineapple/include/asterisk/doxyref.h
team/oej/codename-pineapple/main/asterisk.c
team/oej/codename-pineapple/main/channel.c
team/oej/codename-pineapple/main/manager.c
team/oej/codename-pineapple/main/pbx.c
team/oej/codename-pineapple/main/rtp.c
team/oej/codename-pineapple/main/translate.c
Propchange: team/oej/codename-pineapple/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.
Propchange: team/oej/codename-pineapple/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/oej/codename-pineapple/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Sat Nov 25 06:03:46 2006
@@ -1,1 +1,1 @@
-/trunk:1-47882
+/trunk:1-48005
Modified: team/oej/codename-pineapple/CHANGES
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/CHANGES?view=diff&rev=48006&r1=48005&r2=48006
==============================================================================
--- team/oej/codename-pineapple/CHANGES (original)
+++ team/oej/codename-pineapple/CHANGES Sat Nov 25 06:03:46 2006
@@ -61,3 +61,5 @@
* CID matching information is now shown when doing 'dialplan show'.
* app_queue now has a 'loose' option which is almost exactly like 'strict' except it
does not count paused queue members as unavailable.
+ * Added maxfiles option to options section of asterisk.conf which allows you to specify
+ what Asterisk should set as the maximum number of open files when it loads.
Modified: team/oej/codename-pineapple/channels/chan_h323.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/chan_h323.c?view=diff&rev=48006&r1=48005&r2=48006
==============================================================================
--- team/oej/codename-pineapple/channels/chan_h323.c (original)
+++ team/oej/codename-pineapple/channels/chan_h323.c Sat Nov 25 06:03:46 2006
@@ -2754,21 +2754,6 @@
h323_tokens_show, "Show all active call tokens",
show_tokens_usage };
-static struct ast_cli_entry cli_h323_debug_deprecated = {
- { "h.323", "debug", NULL },
- h323_do_debug, "Enable H.323 debug",
- debug_usage };
-
-static struct ast_cli_entry cli_h323_trace_deprecated = {
- { "h.323", "trace", NULL },
- h323_do_trace, "Enable H.323 Stack Tracing",
- trace_usage };
-
-static struct ast_cli_entry cli_h323_gk_cycle_deprecated = {
- { "h323", "cycle", "gk", NULL },
- h323_gk_cycle, "Manually re-register with the Gatekeper",
- show_cycle_usage };
-
static struct ast_cli_entry cli_h323[] = {
{ { "h323", "set", "trace", NULL },
h323_do_trace, "Enable H.323 Stack Tracing",
Modified: team/oej/codename-pineapple/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/chan_sip.c?view=diff&rev=48006&r1=48005&r2=48006
==============================================================================
--- team/oej/codename-pineapple/channels/chan_sip.c (original)
+++ team/oej/codename-pineapple/channels/chan_sip.c Sat Nov 25 06:03:46 2006
@@ -558,7 +558,7 @@
static int allow_external_domains; /*!< Accept calls to external SIP domains? */
static int global_callevents; /*!< Whether we send manager events or not */
static int global_t1min; /*!< T1 roundtrip time minimum */
-static int global_autoframing; /*!< ?????????? */
+static int global_autoframing; /*!< Turn autoframing on or off. */
static enum transfermodes global_allowtransfer; /*!< SIP Refer restriction scheme */
/*! \brief Codecs that we support by default: */
@@ -11724,7 +11724,7 @@
/* RFC3261 says we must treat every 1xx response (but not 100)
that we don't recognize as if it was 183.
*/
- if (resp > 100 && resp < 200 && resp != 180 && resp != 183)
+ if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 183)
resp = 183;
/* Any response between 100 and 199 is PROCEEDING */
@@ -11738,6 +11738,7 @@
switch (resp) {
case 100: /* Trying */
+ case 101: /* Dialog establishment */
if (!ast_test_flag(req, SIP_PKT_IGNORE))
sip_cancel_destroy(p);
check_pendings(p);
@@ -12232,6 +12233,7 @@
} else if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
switch(resp) {
case 100: /* 100 Trying */
+ case 101: /* 101 Dialog establishment */
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
break;
@@ -12940,6 +12942,7 @@
respcode = atoi(code);
switch (respcode) {
case 100: /* Trying: */
+ case 101: /* dialog establishment */
/* Don't do anything yet */
break;
case 183: /* Ringing: */
@@ -14453,7 +14456,7 @@
if ((firststate = ast_extension_state(NULL, p->context, p->exten)) < 0) {
- ast_log(LOG_ERROR, "Got SUBSCRIBE for extension %s@%s from %s, but there is no hint for that extension\n", p->exten, p->context, ast_inet_ntoa(p->sa.sin_addr));
+ ast_log(LOG_NOTICE, "Got SUBSCRIBE for extension %s@%s from %s, but there is no hint for that extension.\n", p->exten, p->context, ast_inet_ntoa(p->sa.sin_addr));
transmit_response(p, "404 Not found", req);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
return 0;
@@ -14546,7 +14549,6 @@
{
/* Called with p->lock held, as well as p->owner->lock if appropriate, keeping things
relatively static */
- struct sip_request resp;
const char *cmd;
const char *cseq;
const char *useragent;
@@ -14557,9 +14559,6 @@
int debug = sip_debug_test_pvt(p);
char *e;
int error = 0;
-
- /* Clear out potential response */
- memset(&resp, 0, sizeof(resp));
/* Get Method and Cseq */
cseq = get_header(req, "Cseq");
@@ -17116,6 +17115,7 @@
sip_destroy(iterator->call);
}
ASTOBJ_UNLOCK(iterator);
+
} while(0));
/* Then, actually destroy users and registry */
Modified: team/oej/codename-pineapple/doc/asterisk-conf.txt
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/doc/asterisk-conf.txt?view=diff&rev=48006&r1=48005&r2=48006
==============================================================================
--- team/oej/codename-pineapple/doc/asterisk-conf.txt (original)
+++ team/oej/codename-pineapple/doc/asterisk-conf.txt Sat Nov 25 06:03:46 2006
@@ -69,6 +69,7 @@
; when off, sound files are searched as <path>/<lang>/<file>
; when on, sound files are search as <lang>/<path>/<file>
; (only affects relative paths for sound files)
+maxlimit = <value> ; Maximum number open files for the Asterisk process
[files]
; Changing the following lines may compromise your security
Modified: team/oej/codename-pineapple/include/asterisk/causes.h
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/include/asterisk/causes.h?view=diff&rev=48006&r1=48005&r2=48006
==============================================================================
--- team/oej/codename-pineapple/include/asterisk/causes.h (original)
+++ team/oej/codename-pineapple/include/asterisk/causes.h Sat Nov 25 06:03:46 2006
@@ -22,6 +22,68 @@
#ifndef _ASTERISK_CAUSES_H
#define _ASTERISK_CAUSES_H
+
+/*! \page AstCauses Hangup Causes for Asterisk
+
+The Asterisk hangup causes are delivered to the dialplan in the
+${HANGUPCAUSE} channel variable after a call (after execution
+of "dial").
+
+In SIP, we have a conversion table to convert between SIP
+return codes and Q.931 both ways. This is to improve SIP/ISDN
+compatibility.
+
+These are the current codes, based on the Q.931
+specification:
+
+ - AST_CAUSE_UNALLOCATED 1
+ - AST_CAUSE_NO_ROUTE_TRANSIT_NET 2
+ - AST_CAUSE_NO_ROUTE_DESTINATION 3
+ - AST_CAUSE_CHANNEL_UNACCEPTABLE 6
+ - AST_CAUSE_CALL_AWARDED_DELIVERED 7
+ - AST_CAUSE_NORMAL_CLEARING 16
+ - AST_CAUSE_USER_BUSY 17
+ - AST_CAUSE_NO_USER_RESPONSE 18
+ - AST_CAUSE_NO_ANSWER 19
+ - AST_CAUSE_CALL_REJECTED 21
+ - AST_CAUSE_NUMBER_CHANGED 22
+ - AST_CAUSE_DESTINATION_OUT_OF_ORDER 27
+ - AST_CAUSE_INVALID_NUMBER_FORMAT 28
+ - AST_CAUSE_FACILITY_REJECTED 29
+ - AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY 30
+ - AST_CAUSE_NORMAL_UNSPECIFIED 31
+ - AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34
+ - AST_CAUSE_NETWORK_OUT_OF_ORDER 38
+ - AST_CAUSE_NORMAL_TEMPORARY_FAILURE 41
+ - AST_CAUSE_SWITCH_CONGESTION 42
+ - AST_CAUSE_ACCESS_INFO_DISCARDED 43
+ - AST_CAUSE_REQUESTED_CHAN_UNAVAIL 44
+ - AST_CAUSE_PRE_EMPTED 45
+ - AST_CAUSE_FACILITY_NOT_SUBSCRIBED 50
+ - AST_CAUSE_OUTGOING_CALL_BARRED 52
+ - AST_CAUSE_INCOMING_CALL_BARRED 54
+ - AST_CAUSE_BEARERCAPABILITY_NOTAUTH 57
+ - AST_CAUSE_BEARERCAPABILITY_NOTAVAIL 58
+ - AST_CAUSE_BEARERCAPABILITY_NOTIMPL 65
+ - AST_CAUSE_CHAN_NOT_IMPLEMENTED 66
+ - AST_CAUSE_FACILITY_NOT_IMPLEMENTED 69
+ - AST_CAUSE_INVALID_CALL_REFERENCE 81
+ - AST_CAUSE_INCOMPATIBLE_DESTINATION 88
+ - AST_CAUSE_INVALID_MSG_UNSPECIFIED 95
+ - AST_CAUSE_MANDATORY_IE_MISSING 96
+ - AST_CAUSE_MESSAGE_TYPE_NONEXIST 97
+ - AST_CAUSE_WRONG_MESSAGE 98
+ - AST_CAUSE_IE_NONEXIST 99
+ - AST_CAUSE_INVALID_IE_CONTENTS 100
+ - AST_CAUSE_WRONG_CALL_STATE 101
+ - AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE 102
+ - AST_CAUSE_MANDATORY_IE_LENGTH_ERROR 103
+ - AST_CAUSE_PROTOCOL_ERROR 111
+ - AST_CAUSE_INTERWORKING 127
+
+For more information:
+- \ref app_dial.c
+*/
/* Causes for disconnection (from Q.931) */
#define AST_CAUSE_UNALLOCATED 1
Modified: team/oej/codename-pineapple/include/asterisk/channel.h
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/include/asterisk/channel.h?view=diff&rev=48006&r1=48005&r2=48006
==============================================================================
--- team/oej/codename-pineapple/include/asterisk/channel.h (original)
+++ team/oej/codename-pineapple/include/asterisk/channel.h Sat Nov 25 06:03:46 2006
@@ -84,6 +84,11 @@
*/
+/*! \page AstFileDesc File descriptors
+ Asterisk File descriptors are connected to each channel (see \ref Def_Channel)
+ in the \ref ast_channel structure.
+*/
+
#ifndef _ASTERISK_CHANNEL_H
#define _ASTERISK_CHANNEL_H
@@ -336,8 +341,8 @@
AST_STRING_FIELD(uniqueid); /*!< Unique Channel Identifier */
);
- /*! \brief File descriptor for channel -- Drivers will poll on these file descriptors, so at least one must be non -1. */
- int fds[AST_MAX_FDS];
+ /*! \brief File descriptor for channel -- Drivers will poll on these file descriptors, so at least one must be non -1. See \ref AstFileDesc */
+ int fds[AST_MAX_FDS];
void *music_state; /*!< Music State*/
void *generatordata; /*!< Current generator data if there is any */
@@ -351,12 +356,11 @@
struct ast_channel *masqr; /*!< Who we are masquerading as */
int cdrflags; /*!< Call Detail Record Flags */
- /*! \brief Whether or not we have been hung up... Do not set this value
- directly, use ast_softhangup */
- int _softhangup;
+ int _softhangup; /*!< Whether or not we have been hung up... Do not set this value
+ directly, use ast_softhangup() */
time_t whentohangup; /*!< Non-zero, set to actual time when channel is to be hung up */
pthread_t blocker; /*!< If anyone is blocking, this is them */
- ast_mutex_t lock; /*!< Lock, can be used to lock a channel for some operations */
+ ast_mutex_t lock; /*!< Lock, can be used to lock a channel for some operations - see ast_channel_lock() */
const char *blockproc; /*!< Procedure causing blocking */
const char *appl; /*!< Current application */
@@ -373,7 +377,7 @@
int (*timingfunc)(void *data);
void *timingdata;
- enum ast_channel_state _state; /*!< State of line -- Don't write directly, use ast_setstate */
+ enum ast_channel_state _state; /*!< State of line -- Don't write directly, use ast_setstate() */
int rings; /*!< Number of rings so far */
struct ast_callerid cid; /*!< Caller ID, name, presentation etc */
char dtmfq[AST_MAX_EXTENSION]; /*!< Any/all queued DTMF characters */
@@ -397,17 +401,16 @@
struct ast_channel_monitor *monitor; /*!< Channel monitoring */
- /*! Track the read/written samples for monitor use */
- unsigned long insmpl;
- unsigned long outsmpl;
-
- /* Frames in/out counters. The high bit is a debug mask, so
- * the counter is only in the remaining bits
- */
- unsigned int fin;
- unsigned int fout;
+ unsigned long insmpl; /*!< Track the read/written samples for monitor use */
+ unsigned long outsmpl; /*!< Track the read/written samples for monitor use */
+
+ unsigned int fin; /*!< Frames in counters. The high bit is a debug mask, so
+ * the counter is only in the remaining bits */
+ unsigned int fout; /*!< Frames out counters. The high bit is a debug mask, so
+ * the counter is only in the remaining bits */
int hangupcause; /*!< Why is the channel hanged up. See causes.h */
- struct varshead varshead; /*!< A linked list for channel variables */
+ struct varshead varshead; /*!< A linked list for channel variables
+ (see \ref AstChanVar ) */
ast_group_t callgroup; /*!< Call group for call pickups */
ast_group_t pickupgroup; /*!< Pickup group - which calls groups can be picked up? */
unsigned int flags; /*!< channel flags of AST_FLAG_ type */
Modified: team/oej/codename-pineapple/include/asterisk/doxyref.h
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/include/asterisk/doxyref.h?view=diff&rev=48006&r1=48005&r2=48006
==============================================================================
--- team/oej/codename-pineapple/include/asterisk/doxyref.h (original)
+++ team/oej/codename-pineapple/include/asterisk/doxyref.h Sat Nov 25 06:03:46 2006
@@ -135,9 +135,67 @@
* \verbinclude video.txt
*/
-/*! \page AstVar Global channel variables
- * \section globchan Global Channel Variables
+/*! \page AstVar Globally predefined channel variables
+ * \section globchan Globally predefined channel variables
+ *
+ * More and more of these variables are being replaced by dialplan functions.
+ * Some still exist though and some that does still exist needs to move to
+ * dialplan functions.
+ *
+ * See also
+ * - \ref pbx_retrieve_variable()
+ * - \ref AstChanVar
+ *
* \verbinclude channelvariables.txt
+
+ */
+
+/*! \page AstChanVar Asterisk Dialplan Variables
+ * Asterisk Dialplan variables are divided into three groups:
+ * - Predefined global variables, handled by the PBX core
+ * - Global variables, that exist for the duration of the pbx execution
+ * - Channel variables, that exist during a channel
+ *
+ * Global variables are reachable in all channels, all of the time.
+ * Channel variables are only reachable within the channel.
+ *
+ * For more information on the predefined variables, see \ref AstVar
+ *
+ * Global and Channel variables:
+ * - Names are Case insensitive
+ * - Names that start with a character, but are alphanumeric
+ * - Global variables are defined and reached with the GLOBAL() dialplan function
+ * and the set application, like
+ *
+ * exten => 1234,1,set(GLOBAL(myvariable)=tomteluva)
+ *
+ * - \ref func_global.c
+ *
+ * - Channel variables are defined with the set() dialplan application
+ *
+ * exten => 1234,1,set(xmasattribute=tomtegröt)
+ *
+ * - Some channels also supports setting channel variables with the \b setvar=
+ * configuraiton option for a device or line.
+ *
+ * \section AstChanVar_globalvars Global Variables
+ * Global variables can also be set in the [globals] section of extensions.conf. The
+ * setting \b clearglobalvars in extensions.conf [general] section affects whether
+ * or not the global variables defined in \b globals are reset at dialplan reload.
+ *
+ * There are CLI commands to change and read global variables. This can be handy
+ * to reset counters at midnight from an external script.
+ *
+ * \section AstChanVar_devnotes Developer notes
+ * Variable handling is managed within \ref pbx.c
+ * You need to include pbx.h to reach these functions.
+ * - \ref pbx_builtin_setvar_helper()
+ * - \ref pbx_builtin_getvar_helper()
+ *
+ * The variables is a linked list stored in the channel data structure
+ * with the list starting at varshead in struct ast_channel
+ *
+ *
*/
/*! \page AstENUM ENUM
@@ -209,7 +267,7 @@
* \verbinclude features.conf.sample
*/
-/*! \page Config_followme followme.conf
+/*! \page Config_followme Followme: An application for simple follow-me calls
* \section followmeconf Followme.conf
* - See app_followme.c
* \verbinclude followme.conf.sample
Modified: team/oej/codename-pineapple/main/asterisk.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/main/asterisk.c?view=diff&rev=48006&r1=48005&r2=48006
==============================================================================
--- team/oej/codename-pineapple/main/asterisk.c (original)
+++ team/oej/codename-pineapple/main/asterisk.c Sat Nov 25 06:03:46 2006
@@ -1011,6 +1011,29 @@
if (n == 0 && option_debug)
printf("Huh? Child handler, but nobody there?\n");
signal(sig, child_handler);
+}
+
+/*! \brief Set maximum open files */
+static void set_ulimit(int value)
+{
+ struct rlimit l = {0, 0};
+
+ if (value <= 0) {
+ ast_log(LOG_WARNING, "Unable to change max files open to invalid value %i\n",value);
+ return;
+ }
+
+ l.rlim_cur = value;
+ l.rlim_max = value;
+
+ if (setrlimit(RLIMIT_NOFILE, &l)) {
+ ast_log(LOG_WARNING, "Unable to disable core size resource limit: %s\n",strerror(errno));
+ return;
+ }
+
+ ast_log(LOG_NOTICE, "Setting max files open to %d\n",value);
+
+ return;
}
/*! \brief Set an X-term or screen title */
@@ -2293,6 +2316,9 @@
} else if ((sscanf(v->value, "%lf", &option_maxload) != 1) || (option_maxload < 0.0)) {
option_maxload = 0.0;
}
+ /* Set the maximum amount of open files */
+ } else if (!strcasecmp(v->name, "maxfiles")) {
+ set_ulimit(atoi(v->value));
/* What user to run as */
} else if (!strcasecmp(v->name, "runuser")) {
ast_copy_string(ast_config_AST_RUN_USER, v->value, sizeof(ast_config_AST_RUN_USER));
Modified: team/oej/codename-pineapple/main/channel.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/main/channel.c?view=diff&rev=48006&r1=48005&r2=48006
==============================================================================
--- team/oej/codename-pineapple/main/channel.c (original)
+++ team/oej/codename-pineapple/main/channel.c Sat Nov 25 06:03:46 2006
@@ -1027,6 +1027,7 @@
free(cid->cid_ani);
if (cid->cid_rdnis)
free(cid->cid_rdnis);
+ cid->cid_dnid = cid->cid_num = cid->cid_name = cid->cid_ani = cid->cid_rdnis = NULL;
}
/*! \brief Free a channel structure */
Modified: team/oej/codename-pineapple/main/manager.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/main/manager.c?view=diff&rev=48006&r1=48005&r2=48006
==============================================================================
--- team/oej/codename-pineapple/main/manager.c (original)
+++ team/oej/codename-pineapple/main/manager.c Sat Nov 25 06:03:46 2006
@@ -1981,6 +1981,10 @@
return 0;
}
+ if (!s->authenticated && strcasecmp(action, "Login") && strcasecmp(action, "Logoff") && strcasecmp(action, "Challenge")) {
+ astman_send_error(s, m, "Permission denied");
+ return 0;
+ }
/* XXX should we protect the list navigation ? */
for (tmp = first_action ; tmp; tmp = tmp->next) {
if (!strcasecmp(action, tmp->action)) {
Modified: team/oej/codename-pineapple/main/pbx.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/main/pbx.c?view=diff&rev=48006&r1=48005&r2=48006
==============================================================================
--- team/oej/codename-pineapple/main/pbx.c (original)
+++ team/oej/codename-pineapple/main/pbx.c Sat Nov 25 06:03:46 2006
@@ -1094,9 +1094,12 @@
return ret;
}
-/*! \brief pbx_retrieve_variable: Support for Asterisk built-in variables and
- functions in the dialplan
- ---*/
+/*! \brief Support for Asterisk built-in variables and functions in the dialplan
+
+\note See also
+ - \ref AstVar Channel variables
+ - \ref AstCauses The HANGUPCAUSE variable
+ */
void pbx_retrieve_variable(struct ast_channel *c, const char *var, char **ret, char *workspace, int workspacelen, struct varshead *headp)
{
const char not_found = '\0';
Modified: team/oej/codename-pineapple/main/rtp.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/main/rtp.c?view=diff&rev=48006&r1=48005&r2=48006
==============================================================================
--- team/oej/codename-pineapple/main/rtp.c (original)
+++ team/oej/codename-pineapple/main/rtp.c Sat Nov 25 06:03:46 2006
@@ -1705,7 +1705,7 @@
if (isAstFormat &&
(code == AST_FORMAT_G726_AAL2) &&
(options & AST_RTP_OPT_G726_NONSTANDARD))
- return "AAL2-G726-32";
+ return "G726-32";
else
return mimeTypes[i].subtype;
}
@@ -2474,6 +2474,7 @@
return 0;
}
+/*! \brief Write RTP packet with audio or video media frames into UDP packet */
static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec)
{
unsigned char *rtpheader;
@@ -2659,11 +2660,10 @@
ast_rtp_raw_write(rtp, f, codec);
} else {
/* Don't buffer outgoing frames; send them one-per-packet: */
- if (_f->offset < hdrlen) {
- f = ast_frdup(_f);
- } else {
+ if (_f->offset < hdrlen)
+ f = ast_frdup(_f); /*! \bug XXX this might never be free'd. Why do we do this? */
+ else
f = _f;
- }
ast_rtp_raw_write(rtp, f, codec);
}
@@ -2850,7 +2850,7 @@
return AST_BRIDGE_FAILED;
}
-/*! \brief P2P RTP/RTCP Callback */
+/*! \brief peer 2 peer RTP mode RTP/RTCP Callback */
static int p2p_rtp_callback(int *id, int fd, short events, void *cbdata)
{
int res = 0, hdrlen = 12;
@@ -2951,8 +2951,13 @@
return 0;
}
-/*! \brief Bridge loop for partial native bridge (packet2packet) */
-static enum ast_bridge_result bridge_p2p_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp *p0, struct ast_rtp *p1, struct ast_rtp *vp0, struct ast_rtp *vp1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
+/*! \brief Bridge loop for partial native bridge (packet2packet)
+
+ In p2p mode, Asterisk is a very basic RTP proxy, just forwarding whatever
+ rtp/rtcp we get in to the channel.
+ \note this currently only works for Audio
+*/
+static enum ast_bridge_result bridge_p2p_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp *p0, struct ast_rtp *p1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
{
struct ast_frame *fr = NULL;
struct ast_channel *who = NULL, *other = NULL, *cs[3] = {NULL, };
@@ -2965,12 +2970,6 @@
p0->bridged = p1;
ast_clear_flag(p1, FLAG_P2P_SENT_MARK);
p1->bridged = p0;
- if (vp0) {
- ast_clear_flag(vp0, FLAG_P2P_SENT_MARK);
- vp0->bridged = vp1;
- ast_clear_flag(vp1, FLAG_P2P_SENT_MARK);
- vp1->bridged = vp0;
- }
/* Activate callback modes if possible */
p0_callback = p2p_callback_enable(c0, p0, &p0_iod[0]);
@@ -3032,22 +3031,12 @@
p1_callback = p2p_callback_disable(c1, p1, &p1_iod[0]);
p0->bridged = NULL;
p1->bridged = NULL;
- if (vp0) {
- vp0->bridged = NULL;
- vp1->bridged = NULL;
- }
} else if (fr->subclass == AST_CONTROL_UNHOLD) {
/* If we are off hold, then go back to callback mode and P2P bridging */
ast_clear_flag(p0, FLAG_P2P_SENT_MARK);
p0->bridged = p1;
ast_clear_flag(p1, FLAG_P2P_SENT_MARK);
p1->bridged = p0;
- if (vp0) {
- ast_clear_flag(vp0, FLAG_P2P_SENT_MARK);
- vp0->bridged = vp1;
- ast_clear_flag(vp1, FLAG_P2P_SENT_MARK);
- vp1->bridged = vp0;
- }
p0_callback = p2p_callback_enable(c0, p0, &p0_iod[0]);
p1_callback = p2p_callback_enable(c1, p1, &p1_iod[0]);
}
@@ -3085,10 +3074,6 @@
/* Break out of the direct bridge */
p0->bridged = NULL;
p1->bridged = NULL;
- if (vp0) {
- vp0->bridged = NULL;
- vp1->bridged = NULL;
- }
return res;
}
@@ -3188,7 +3173,7 @@
}
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Packet2Packet bridging %s and %s\n", c0->name, c1->name);
- res = bridge_p2p_loop(c0, c1, p0, p1, vp0, vp1, timeoutms, flags, fo, rc, pvt0, pvt1);
+ res = bridge_p2p_loop(c0, c1, p0, p1, timeoutms, flags, fo, rc, pvt0, pvt1);
} else {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name);
Modified: team/oej/codename-pineapple/main/translate.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/main/translate.c?view=diff&rev=48006&r1=48005&r2=48006
==============================================================================
--- team/oej/codename-pineapple/main/translate.c (original)
+++ team/oej/codename-pineapple/main/translate.c Sat Nov 25 06:03:46 2006
@@ -172,6 +172,7 @@
}
l = plc_fillin(pvt->plc, dst + pvt->samples, l);
pvt->samples += l;
+ pvt->datalen = pvt->samples * 2; /* SLIN has 2bytes for 1sample */
}
return 0;
}
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