[svn-commits] oej: branch oej/videocaps r48211 - in
/team/oej/videocaps: ./ agi/ apps/ chan...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Sat Dec 2 11:54:42 MST 2006
Author: oej
Date: Sat Dec 2 12:54:41 2006
New Revision: 48211
URL: http://svn.digium.com/view/asterisk?view=rev&rev=48211
Log:
Update to trunk
Removed:
team/oej/videocaps/main/coef_in.h
team/oej/videocaps/main/coef_out.h
Modified:
team/oej/videocaps/ (props changed)
team/oej/videocaps/Makefile
team/oej/videocaps/agi/Makefile
team/oej/videocaps/agi/eagi-sphinx-test.c
team/oej/videocaps/agi/eagi-test.c
team/oej/videocaps/apps/app_dial.c
team/oej/videocaps/apps/app_sms.c
team/oej/videocaps/apps/app_voicemail.c
team/oej/videocaps/channels/chan_gtalk.c
team/oej/videocaps/channels/chan_iax2.c
team/oej/videocaps/channels/chan_phone.c
team/oej/videocaps/channels/chan_sip.c
team/oej/videocaps/channels/chan_zap.c
team/oej/videocaps/codecs/codec_zap.c
team/oej/videocaps/configs/extensions.conf.sample
team/oej/videocaps/configs/http.conf.sample
team/oej/videocaps/configs/sip.conf.sample
team/oej/videocaps/configure
team/oej/videocaps/configure.ac
team/oej/videocaps/contrib/asterisk-ng-doxygen
team/oej/videocaps/doc/asterisk-conf.txt
team/oej/videocaps/doc/manager.txt
team/oej/videocaps/funcs/func_cdr.c
team/oej/videocaps/include/asterisk/causes.h
team/oej/videocaps/include/asterisk/channel.h
team/oej/videocaps/include/asterisk/devicestate.h
team/oej/videocaps/include/asterisk/doxyref.h
team/oej/videocaps/include/asterisk/pbx.h
team/oej/videocaps/include/asterisk/rtp.h
team/oej/videocaps/include/asterisk/threadstorage.h
team/oej/videocaps/include/asterisk/utils.h
team/oej/videocaps/main/callerid.c
team/oej/videocaps/main/cdr.c
team/oej/videocaps/main/channel.c
team/oej/videocaps/main/cli.c
team/oej/videocaps/main/devicestate.c
team/oej/videocaps/main/fskmodem.c
team/oej/videocaps/main/http.c
team/oej/videocaps/main/manager.c
team/oej/videocaps/main/pbx.c
team/oej/videocaps/main/rtp.c
team/oej/videocaps/main/tdd.c
team/oej/videocaps/main/translate.c
team/oej/videocaps/main/utils.c
team/oej/videocaps/makeopts.in
team/oej/videocaps/pbx/pbx_spool.c
team/oej/videocaps/res/res_features.c
team/oej/videocaps/res/res_musiconhold.c
team/oej/videocaps/sounds/Makefile
Propchange: team/oej/videocaps/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.
Propchange: team/oej/videocaps/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/oej/videocaps/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Sat Dec 2 12:54:41 2006
@@ -1,1 +1,1 @@
-/trunk:1-47956
+/trunk:1-48207
Modified: team/oej/videocaps/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/Makefile?view=diff&rev=48211&r1=48210&r2=48211
==============================================================================
--- team/oej/videocaps/Makefile (original)
+++ team/oej/videocaps/Makefile Sat Dec 2 12:54:41 2006
@@ -37,6 +37,11 @@
export ASTVARLIBDIR
export ASTDATADIR
export ASTLOGDIR
+export ASTLIBDIR
+export ASTMANDIR
+export ASTHEADERDIR
+export ASTBINDIR
+export ASTSBINDIR
export AGI_DIR
export ASTCONFPATH
export NOISY_BUILD
@@ -52,6 +57,7 @@
export PROC
export SOLINK
export STRIP
+export DOWNLOAD
# even though we could use '-include makeopts' here, use a wildcard
# lookup anyway, so that make won't try to build makeopts if it doesn't
@@ -268,14 +274,14 @@
@echo " + Asterisk has successfully been built, and +"
@echo " + can be installed by running: +"
@echo " + +"
- @echo " + make install +"
+ @echo " + $(MAKE) install +"
@echo " +-------------------------------------------+"
_all: cleantest $(SUBDIRS)
makeopts: configure
@echo "****"
- @echo "**** The configure script must be executed before running 'make'."
+ @echo "**** The configure script must be executed before running '$(MAKE)'."
@echo "****"
@exit 1
Modified: team/oej/videocaps/agi/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/agi/Makefile?view=diff&rev=48211&r1=48210&r2=48211
==============================================================================
--- team/oej/videocaps/agi/Makefile (original)
+++ team/oej/videocaps/agi/Makefile Sat Dec 2 12:54:41 2006
@@ -21,7 +21,7 @@
include $(ASTTOPDIR)/Makefile.rules
-all: ##$(AGIS)
+all: #$(AGIS)
strcompat.c: ../main/strcompat.c
@cp $< $@
Modified: team/oej/videocaps/agi/eagi-sphinx-test.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/agi/eagi-sphinx-test.c?view=diff&rev=48211&r1=48210&r2=48211
==============================================================================
--- team/oej/videocaps/agi/eagi-sphinx-test.c (original)
+++ team/oej/videocaps/agi/eagi-sphinx-test.c Sat Dec 2 12:54:41 2006
@@ -4,6 +4,15 @@
* This code is released into public domain
* without any warranty of any kind.
*
+ */
+
+/*! \file
+ * Extended AGI test application
+ *
+ * This code is released into public domain
+ * without any warranty of any kind.
+ *
+ * \ingroup agi
*/
#include <stdio.h>
Modified: team/oej/videocaps/agi/eagi-test.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/agi/eagi-test.c?view=diff&rev=48211&r1=48210&r2=48211
==============================================================================
--- team/oej/videocaps/agi/eagi-test.c (original)
+++ team/oej/videocaps/agi/eagi-test.c Sat Dec 2 12:54:41 2006
@@ -17,6 +17,15 @@
#include "asterisk/compat.h"
#define AUDIO_FILENO (STDERR_FILENO + 1)
+
+/*! \file
+ * Extended AGI test application
+ *
+ * This code is released into the public domain
+ * with no warranty of any kind
+ *
+ * \ingroup agi
+ */
static int read_environment(void)
{
Modified: team/oej/videocaps/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/apps/app_dial.c?view=diff&rev=48211&r1=48210&r2=48211
==============================================================================
--- team/oej/videocaps/apps/app_dial.c (original)
+++ team/oej/videocaps/apps/app_dial.c Sat Dec 2 12:54:41 2006
@@ -1441,6 +1441,7 @@
if (ast_test_flag(outgoing, OPT_MUSICBACK)) {
moh = 1;
ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
+ ast_indicate(chan, AST_CONTROL_PROGRESS);
} else if (ast_test_flag(outgoing, OPT_RINGBACK)) {
ast_indicate(chan, AST_CONTROL_RINGING);
sentringing++;
Modified: team/oej/videocaps/apps/app_sms.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/apps/app_sms.c?view=diff&rev=48211&r1=48210&r2=48211
==============================================================================
--- team/oej/videocaps/apps/app_sms.c (original)
+++ team/oej/videocaps/apps/app_sms.c Sat Dec 2 12:54:41 2006
@@ -119,59 +119,59 @@
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
};
-#define SMSLEN 160 /* max SMS length */
+#define SMSLEN 160 /*!< max SMS length */
typedef struct sms_s
{
- unsigned char hangup; /* we are done... */
- unsigned char err; /* set for any errors */
- unsigned char smsc:1; /* we are SMSC */
- unsigned char rx:1; /* this is a received message */
- char queue[30]; /* queue name */
- char oa[20]; /* originating address */
- char da[20]; /* destination address */
- time_t scts; /* time stamp, UTC */
- unsigned char pid; /* protocol ID */
- unsigned char dcs; /* data coding scheme */
- short mr; /* message reference - actually a byte, but usde -1 for not set */
- int udl; /* user data length */
- int udhl; /* user data header length */
- unsigned char srr:1; /* Status Report request */
- unsigned char udhi:1; /* User Data Header required, even if length 0 */
- unsigned char rp:1; /* Reply Path */
- unsigned int vp; /* validity period in minutes, 0 for not set */
- unsigned short ud[SMSLEN]; /* user data (message), UCS-2 coded */
- unsigned char udh[SMSLEN]; /* user data header */
- char cli[20]; /* caller ID */
- unsigned char ophase; /* phase (0-79) for 0 and 1 frequencies (1300Hz and 2100Hz) */
- unsigned char ophasep; /* phase (0-79) for 1200 bps */
- unsigned char obyte; /* byte being sent */
- unsigned int opause; /* silent pause before sending (in sample periods) */
- unsigned char obitp; /* bit in byte */
- unsigned char osync; /* sync bits to send */
- unsigned char obytep; /* byte in data */
- unsigned char obyten; /* bytes in data */
- unsigned char omsg[256]; /* data buffer (out) */
- unsigned char imsg[200]; /* data buffer (in) */
+ unsigned char hangup; /*!< we are done... */
+ unsigned char err; /*!< set for any errors */
+ unsigned char smsc:1; /*!< we are SMSC */
+ unsigned char rx:1; /*!< this is a received message */
+ char queue[30]; /*!< queue name */
+ char oa[20]; /*!< originating address */
+ char da[20]; /*!< destination address */
+ time_t scts; /*!< time stamp, UTC */
+ unsigned char pid; /*!< protocol ID */
+ unsigned char dcs; /*!< data coding scheme */
+ short mr; /*!< message reference - actually a byte, but usde -1 for not set */
+ int udl; /*!< user data length */
+ int udhl; /*!< user data header length */
+ unsigned char srr:1; /*!< Status Report request */
+ unsigned char udhi:1; /*!< User Data Header required, even if length 0 */
+ unsigned char rp:1; /*!< Reply Path */
+ unsigned int vp; /*!< validity period in minutes, 0 for not set */
+ unsigned short ud[SMSLEN]; /*!< user data (message), UCS-2 coded */
+ unsigned char udh[SMSLEN]; /*!< user data header */
+ char cli[20]; /*!< caller ID */
+ unsigned char ophase; /*!< phase (0-79) for 0 and 1 frequencies (1300Hz and 2100Hz) */
+ unsigned char ophasep; /*!< phase (0-79) for 1200 bps */
+ unsigned char obyte; /*!< byte being sent */
+ unsigned int opause; /*!< silent pause before sending (in sample periods) */
+ unsigned char obitp; /*!< bit in byte */
+ unsigned char osync; /*!< sync bits to send */
+ unsigned char obytep; /*!< byte in data */
+ unsigned char obyten; /*!< bytes in data */
+ unsigned char omsg[256]; /*!< data buffer (out) */
+ unsigned char imsg[200]; /*!< data buffer (in) */
signed long long ims0,
imc0,
ims1,
- imc1; /* magnitude averages sin/cos 0/1 */
+ imc1; /*!< magnitude averages sin/cos 0/1 */
unsigned int idle;
- unsigned short imag; /* signal level */
+ unsigned short imag; /*!< signal level */
unsigned char ips0,
ips1,
ipc0,
- ipc1; /* phase sin/cos 0/1 */
- unsigned char ibitl; /* last bit */
- unsigned char ibitc; /* bit run length count */
- unsigned char iphasep; /* bit phase (0-79) for 1200 bps */
- unsigned char ibitn; /* bit number in byte being received */
- unsigned char ibytev; /* byte value being received */
- unsigned char ibytep; /* byte pointer in messafe */
- unsigned char ibytec; /* byte checksum for message */
- unsigned char ierr; /* error flag */
- unsigned char ibith; /* history of last bits */
+ ipc1; /*!< phase sin/cos 0/1 */
+ unsigned char ibitl; /*!< last bit */
+ unsigned char ibitc; /*!< bit run length count */
+ unsigned char iphasep; /*!< bit phase (0-79) for 1200 bps */
+ unsigned char ibitn; /*!< bit number in byte being received */
+ unsigned char ibytev; /*!< byte value being received */
+ unsigned char ibytep; /*!< byte pointer in messafe */
+ unsigned char ibytec; /*!< byte checksum for message */
+ unsigned char ierr; /*!< error flag */
+ unsigned char ibith; /*!< history of last bits */
unsigned char ibitt; /* total of 1's in last 3 bites */
/* more to go here */
} sms_t;
Modified: team/oej/videocaps/apps/app_voicemail.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/apps/app_voicemail.c?view=diff&rev=48211&r1=48210&r2=48211
==============================================================================
--- team/oej/videocaps/apps/app_voicemail.c (original)
+++ team/oej/videocaps/apps/app_voicemail.c Sat Dec 2 12:54:41 2006
@@ -4673,7 +4673,7 @@
if(option_debug > 2)
ast_log(LOG_DEBUG,"Before init_mailstream, user is %s\n",vmu->imapuser);
ret = init_mailstream(vms, box);
- if (ret != 0) {
+ if (ret != 0 || !vms->mailstream) {
ast_log (LOG_ERROR,"Could not initialize mailstream\n");
return -1;
}
@@ -6970,22 +6970,13 @@
}
}
AST_LIST_TRAVERSE(&users, vmu, list) {
- char dirname[256];
- DIR *vmdir;
- struct dirent *vment;
- int vmcount = 0;
- char count[12];
+ int newmsgs = 0, oldmsgs = 0;
+ char count[12], tmp[256] = "";
if ((argc == 3) || ((argc == 5) && !strcmp(argv[4],vmu->context))) {
- make_dir(dirname, 255, vmu->context, vmu->mailbox, "INBOX");
- if ((vmdir = opendir(dirname))) {
- /* No matter what the format of VM, there will always be a .txt file for each message. */
- while ((vment = readdir(vmdir)))
- if (strlen(vment->d_name) > 7 && !strncmp(vment->d_name + 7,".txt",4))
- vmcount++;
- closedir(vmdir);
- }
- snprintf(count,sizeof(count),"%d",vmcount);
+ snprintf(tmp, sizeof(tmp), "%s@%s", vmu->mailbox, ast_strlen_zero(vmu->context) ? "default" : vmu->context);
+ inboxcount(tmp, &newmsgs, &oldmsgs);
+ snprintf(count,sizeof(count),"%d",newmsgs);
ast_cli(fd, output_format, vmu->context, vmu->mailbox, vmu->fullname, vmu->zonetag, count);
}
}
@@ -8481,15 +8472,16 @@
if(option_debug > 3)
ast_log(LOG_DEBUG, "Entering callback mm_login\n");
- ast_copy_string(user, mb->user,sizeof(user));
+
+ ast_copy_string(user, mb->user, MAILTMPLEN);
/* We should only do this when necessary */
if (!ast_strlen_zero(authpassword)) {
- ast_copy_string(pwd, authpassword, sizeof(pwd));
+ ast_copy_string(pwd, authpassword, MAILTMPLEN);
} else {
AST_LIST_TRAVERSE(&users, vmu, list) {
if(!strcasecmp(mb->user, vmu->imapuser)) {
- ast_copy_string(pwd, vmu->imappassword, sizeof(pwd));
+ ast_copy_string(pwd, vmu->imappassword, MAILTMPLEN);
break;
}
}
Modified: team/oej/videocaps/channels/chan_gtalk.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/channels/chan_gtalk.c?view=diff&rev=48211&r1=48210&r2=48211
==============================================================================
--- team/oej/videocaps/channels/chan_gtalk.c (original)
+++ team/oej/videocaps/channels/chan_gtalk.c Sat Dec 2 12:54:41 2006
@@ -163,7 +163,6 @@
};
static const char desc[] = "Gtalk Channel";
-static const char type[] = "Gtalk";
static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
@@ -192,7 +191,7 @@
/*! \brief PBX interface structure for channel registration */
static const struct ast_channel_tech gtalk_tech = {
- .type = type,
+ .type = "Gtalk",
.description = "Gtalk Channel Driver",
.capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
.requester = gtalk_request,
@@ -220,7 +219,7 @@
/*! \brief RTP driver interface */
static struct ast_rtp_protocol gtalk_rtp = {
- type: "gtalk",
+ type: "Gtalk",
get_rtp_info: gtalk_get_rtp_peer,
set_rtp_peer: gtalk_set_rtp_peer,
get_codec: gtalk_get_codec,
@@ -921,10 +920,12 @@
fmt = ast_best_codec(tmp->nativeformats);
if (i->rtp) {
+ ast_rtp_setstun(i->rtp, 1);
tmp->fds[0] = ast_rtp_fd(i->rtp);
tmp->fds[1] = ast_rtcp_fd(i->rtp);
}
if (i->vrtp) {
+ ast_rtp_setstun(i->rtp, 1);
tmp->fds[2] = ast_rtp_fd(i->vrtp);
tmp->fds[3] = ast_rtcp_fd(i->vrtp);
}
@@ -1790,7 +1791,7 @@
/* Make sure we can register our channel type */
if (ast_channel_register(>alk_tech)) {
- ast_log(LOG_ERROR, "Unable to register channel class %s\n", type);
+ ast_log(LOG_ERROR, "Unable to register channel class %s\n", gtalk_tech.type);
return -1;
}
return 0;
Modified: team/oej/videocaps/channels/chan_iax2.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/channels/chan_iax2.c?view=diff&rev=48211&r1=48210&r2=48211
==============================================================================
--- team/oej/videocaps/channels/chan_iax2.c (original)
+++ team/oej/videocaps/channels/chan_iax2.c Sat Dec 2 12:54:41 2006
@@ -1195,6 +1195,20 @@
return res;
}
+/*!
+ * \todo XXX Note that this function contains a very expensive operation that
+ * happens for *every* incoming media frame. It iterates through every
+ * possible call number, locking and unlocking each one, to try to match the
+ * incoming frame to an active call. Call numbers can be up to 2^15, 32768.
+ * So, for an call with a local call number of 20000, every incoming audio
+ * frame would require 20000 mutex lock and unlock operations. Ouch.
+ *
+ * It's a shame that IAX2 media frames carry the source call number instead of
+ * the destination call number. If they did, this lookup wouldn't be needed.
+ * However, it's too late to change that now. Instead, we need to come up with
+ * a better way of indexing active calls so that these frequent lookups are not
+ * so expensive.
+ */
static int find_callno(unsigned short callno, unsigned short dcallno, struct sockaddr_in *sin, int new, int lockpeer, int sockfd)
{
int res = 0;
@@ -6974,7 +6988,7 @@
if (!strcmp(ies.called_number, ast_parking_ext())) {
if (iax_park(ast_bridged_channel(iaxs[fr->callno]->owner), iaxs[fr->callno]->owner)) {
ast_log(LOG_WARNING, "Failed to park call on '%s'\n", ast_bridged_channel(iaxs[fr->callno]->owner)->name);
- } else {
+ } else if (ast_bridged_channel(iaxs[fr->callno]->owner)) {
if (option_debug)
ast_log(LOG_DEBUG, "Parked call on '%s'\n", ast_bridged_channel(iaxs[fr->callno]->owner)->name);
}
Modified: team/oej/videocaps/channels/chan_phone.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/channels/chan_phone.c?view=diff&rev=48211&r1=48210&r2=48211
==============================================================================
--- team/oej/videocaps/channels/chan_phone.c (original)
+++ team/oej/videocaps/channels/chan_phone.c Sat Dec 2 12:54:41 2006
@@ -48,11 +48,6 @@
#include <linux/telephony.h>
/* Still use some IXJ specific stuff */
#include <linux/version.h>
-#if LINUX_VERSION_CODE >= KERNEL_VERSION(2,6,0)
-#if LINUX_VERSION_CODE < KERNEL_VERSION(2,6,18)
-# include <linux/compiler.h>
-#endif
-#endif
#include <linux/ixjuser.h>
#include "asterisk/lock.h"
Modified: team/oej/videocaps/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/channels/chan_sip.c?view=diff&rev=48211&r1=48210&r2=48211
==============================================================================
--- team/oej/videocaps/channels/chan_sip.c (original)
+++ team/oej/videocaps/channels/chan_sip.c Sat Dec 2 12:54:41 2006
@@ -980,8 +980,6 @@
time_t lastrtprx; /*!< Last RTP received */
time_t lastrtptx; /*!< Last RTP sent */
int rtptimeout; /*!< RTP timeout time */
- int rtpholdtimeout; /*!< RTP timeout when on hold */
- int rtpkeepalive; /*!< Send RTP packets for keepalive */
struct sockaddr_in recv; /*!< Received as */
struct in_addr ourip; /*!< Our IP */
struct ast_channel *owner; /*!< Who owns us (if we have an owner) */
@@ -2766,21 +2764,25 @@
if (dialog->rtp) {
ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
ast_rtp_setdtmfcompensate(dialog->rtp, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
+ ast_rtp_set_rtptimeout(dialog->rtp, peer->rtptimeout);
+ ast_rtp_set_rtpholdtimeout(dialog->rtp, peer->rtpholdtimeout);
+ ast_rtp_set_rtpkeepalive(dialog->rtp, peer->rtpkeepalive);
+ /* Set Frame packetization */
+ ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
+ dialog->autoframing = peer->autoframing;
}
if (dialog->vrtp) {
ast_rtp_setdtmf(dialog->vrtp, 0);
ast_rtp_setdtmfcompensate(dialog->vrtp, 0);
+ ast_rtp_set_rtptimeout(dialog->vrtp, peer->rtptimeout);
+ ast_rtp_set_rtpholdtimeout(dialog->vrtp, peer->rtpholdtimeout);
+ ast_rtp_set_rtpkeepalive(dialog->vrtp, peer->rtpkeepalive);
}
if (dialog->trtp) {
ast_rtp_setdtmf(dialog->trtp, 0);
ast_rtp_setdtmfcompensate(dialog->trtp, 0);
}
- /* Set Frame packetization */
- if (dialog->rtp) {
- ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
- dialog->autoframing = peer->autoframing;
- }
ast_string_field_set(dialog, peername, peer->username);
ast_string_field_set(dialog, authname, peer->username);
ast_string_field_set(dialog, username, peer->username);
@@ -2818,8 +2820,6 @@
dialog->noncodeccapability &= ~AST_RTP_DTMF;
ast_string_field_set(dialog, context, peer->context);
dialog->rtptimeout = peer->rtptimeout;
- dialog->rtpholdtimeout = peer->rtpholdtimeout;
- dialog->rtpkeepalive = peer->rtpkeepalive;
if (peer->call_limit)
ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
ast_copy_capabilities(&dialog->caps, &peer->caps);
@@ -3157,9 +3157,9 @@
/* Check the list of users only for incoming calls */
if (global_limitonpeers == FALSE && !outgoing && (u = find_user(name, 1))) {
- inuse = &u->inUse;
- call_limit = &u->call_limit;
- inringing = NULL;
+ inuse = &u->inUse;
+ call_limit = &u->call_limit;
+ inringing = NULL;
} else if ( (p = find_peer(ast_strlen_zero(fup->peername) ? name : fup->peername, NULL, 1) ) ) { /* Try to find peer */
inuse = &p->inUse;
call_limit = &p->call_limit;
@@ -3687,15 +3687,12 @@
case AST_FRAME_MODEM:
if (p) {
sip_pvt_lock(p);
- if (p->udptl) {
- if ((ast->_state != AST_STATE_UP) &&
- !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
- !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- transmit_response_with_t38_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
- ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
- }
+ /* UDPTL requires two-way communication, so early media is not needed here.
+ we simply forget the frames if we get modem frames before the bridge is up.
+ Fax will re-transmit.
+ */
+ if (p->udptl && ast->_state != AST_STATE_UP)
res = ast_udptl_write(p->udptl, frame);
- }
sip_pvt_unlock(p);
}
break;
@@ -4513,10 +4510,16 @@
ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
ast_rtp_settos(p->rtp, global_tos_audio);
+ ast_rtp_set_rtptimeout(p->rtp, global_rtptimeout);
+ ast_rtp_set_rtpholdtimeout(p->rtp, global_rtpholdtimeout);
+ ast_rtp_set_rtpkeepalive(p->rtp, global_rtpkeepalive);
if (p->vrtp) {
ast_rtp_settos(p->vrtp, global_tos_video);
ast_rtp_setdtmf(p->vrtp, 0);
ast_rtp_setdtmfcompensate(p->vrtp, 0);
+ ast_rtp_set_rtptimeout(p->vrtp, global_rtptimeout);
+ ast_rtp_set_rtpholdtimeout(p->vrtp, global_rtpholdtimeout);
+ ast_rtp_set_rtpkeepalive(p->vrtp, global_rtpkeepalive);
}
if (p->trtp) {
ast_rtp_settos(p->trtp, global_tos_text);
@@ -4525,9 +4528,6 @@
}
if (p->udptl)
ast_udptl_settos(p->udptl, global_tos_audio);
- p->rtptimeout = global_rtptimeout;
- p->rtpholdtimeout = global_rtpholdtimeout;
- p->rtpkeepalive = global_rtpkeepalive;
p->videoupdate = global_videoupdate;
ast_copy_capabilities(&p->caps, &global_caps);
}
@@ -8121,7 +8121,8 @@
ast_string_field_set(p, domain, r->domain);
ast_string_field_set(p, opaque, r->opaque);
ast_string_field_set(p, qop, r->qop);
- p->noncecount = r->noncecount++;
+ r->noncecount++;
+ p->noncecount = r->noncecount;
memset(digest,0,sizeof(digest));
if(!build_reply_digest(p, sipmethod, digest, sizeof(digest)))
@@ -11233,6 +11234,7 @@
ast_cli(fd, " T1 minimum: %d\n", global_t1min);
ast_cli(fd, " Relax DTMF: %s\n", global_relaxdtmf ? "Yes" : "No");
ast_cli(fd, " Compact SIP headers: %s\n", compactheaders ? "Yes" : "No");
+ ast_cli(fd, " RTP Keepalive: %d %s\n", global_rtpkeepalive, global_rtpkeepalive ? "" : "(Disabled)" );
ast_cli(fd, " RTP Timeout: %d %s\n", global_rtptimeout, global_rtptimeout ? "" : "(Disabled)" );
ast_cli(fd, " RTP Hold Timeout: %d %s\n", global_rtpholdtimeout, global_rtpholdtimeout ? "" : "(Disabled)");
ast_cli(fd, " MWI NOTIFY mime type: %s\n", default_notifymime);
@@ -12634,6 +12636,9 @@
if (bridgepvt->udptl) {
if (p->t38.state == T38_PEER_REINVITE) {
sip_handle_t38_reinvite(bridgepeer, p, 0);
+ ast_rtp_set_rtptimers_onhold(p->rtp);
+ if (p->vrtp)
+ ast_rtp_set_rtptimers_onhold(p->vrtp); /* Turn off RTP timers while we send fax */
} else if (p->t38.state == T38_DISABLED && bridgepeer && (bridgepvt->t38.state == T38_ENABLED)) {
ast_log(LOG_WARNING, "RTP re-inivte after T38 session not handled yet !\n");
/* Insted of this we should somehow re-invite the other side of the bridge to RTP */
@@ -13142,7 +13147,8 @@
if (sipmethod == SIP_INVITE) {
/* First we ACK */
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- ast_log(LOG_WARNING, "INVITE with REPLACEs failed to '%s'\n", get_header(&p->initreq, "From"));
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Got 481 on Invite. Assuming INVITE with REPLACEs failed to '%s'\n", get_header(&p->initreq, "From"));
if (owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
@@ -15279,7 +15285,7 @@
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;
@@ -15372,7 +15378,6 @@
{
/* 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;
@@ -15384,9 +15389,6 @@
char *e;
int error = 0;
- /* Clear out potential response */
- memset(&resp, 0, sizeof(resp));
-
/* Get Method and Cseq */
cseq = get_header(req, "Cseq");
cmd = req->header[0];
@@ -15401,7 +15403,7 @@
error = 1;
}
if (error) {
- if (!p->initreq.header) /* New call */
+ if (!p->initreq.headers) /* New call */
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); /* Make sure we destroy this dialog */
return -1;
}
@@ -15740,52 +15742,66 @@
/*! \brief helper function for the monitoring thread */
-static void check_rtp_timeout(struct sip_pvt *sip, time_t t)
-{
- if (sip->rtp && sip->owner &&
- (sip->owner->_state == AST_STATE_UP) &&
- !sip->redirip.sin_addr.s_addr) {
- if (sip->lastrtptx &&
- sip->rtpkeepalive &&
- (t > sip->lastrtptx + sip->rtpkeepalive)) {
- /* Need to send an empty RTP packet */
- sip->lastrtptx = time(NULL);
- ast_rtp_sendcng(sip->rtp, 0);
- }
- if (sip->lastrtprx &&
- (sip->rtptimeout || sip->rtpholdtimeout) &&
- (t > sip->lastrtprx + sip->rtptimeout)) {
- /* Might be a timeout now -- see if we're on hold */
- struct sockaddr_in sin;
- ast_rtp_get_peer(sip->rtp, &sin);
- if (sin.sin_addr.s_addr ||
- (sip->rtpholdtimeout &&
- (t > sip->lastrtprx + sip->rtpholdtimeout))) {
- /* Needs a hangup */
- if (sip->rtptimeout) {
- while (sip->owner && ast_channel_trylock(sip->owner)) {
- sip_pvt_unlock(sip);
- usleep(1);
- sip_pvt_lock(sip);
- }
- if (sip->owner) {
- if (!(ast_rtp_get_bridged(sip->rtp))) {
- ast_log(LOG_NOTICE,
- "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
- sip->owner->name,
- (long) (t - sip->lastrtprx));
- /* Issue a softhangup */
- ast_softhangup_nolock(sip->owner, AST_SOFTHANGUP_DEV);
- } else
- ast_log(LOG_NOTICE, "'%s' will not be disconnected in %ld seconds because it is directly bridged to another RTP stream\n", sip->owner->name, (long) (t - sip->lastrtprx));
- ast_channel_unlock(sip->owner);
- /* forget the timeouts for this call, since a hangup
- has already been requested and we don't want to
- repeatedly request hangups
- */
- sip->rtptimeout = 0;
- sip->rtpholdtimeout = 0;
- }
+static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
+{
+ /* If we have no RTP or no active owner, no need to check timers */
+ if (!dialog->rtp || !dialog->owner)
+ return;
+ /* If the call is not in UP state or redirected outside Asterisk, no need to check timers */
+ if (dialog->owner->_state != AST_STATE_UP || dialog->redirip.sin_addr.s_addr)
+ return;
+
+ /* If we have no timers set, return now */
+ if (ast_rtp_get_rtpkeepalive(dialog->rtp) == 0 || (ast_rtp_get_rtptimeout(dialog->rtp) == 0 && ast_rtp_get_rtpholdtimeout(dialog->rtp) == 0))
+ return;
+
+ /* Check AUDIO RTP keepalives */
+ if (dialog->lastrtptx && ast_rtp_get_rtpkeepalive(dialog->rtp) &&
+ (t > dialog->lastrtptx + ast_rtp_get_rtpkeepalive(dialog->rtp))) {
+ /* Need to send an empty RTP packet */
+ dialog->lastrtptx = time(NULL);
+ ast_rtp_sendcng(dialog->rtp, 0);
+ }
+
+ /*! \todo Check video RTP keepalives
+
+ Do we need to move the lastrtptx to the RTP structure to have one for audio and one
+ for video? It really does belong to the RTP structure.
+ */
+
+ /* Check AUDIO RTP timers */
+ if (dialog->lastrtprx && (ast_rtp_get_rtptimeout(dialog->rtp) || ast_rtp_get_rtpholdtimeout(dialog->rtp)) &&
+ (t > dialog->lastrtprx + ast_rtp_get_rtptimeout(dialog->rtp))) {
+
+ /* Might be a timeout now -- see if we're on hold */
+ struct sockaddr_in sin;
+ ast_rtp_get_peer(dialog->rtp, &sin);
+ if (sin.sin_addr.s_addr || (ast_rtp_get_rtpholdtimeout(dialog->rtp) &&
+ (t > dialog->lastrtprx + ast_rtp_get_rtpholdtimeout(dialog->rtp)))) {
+ /* Needs a hangup */
+ if (dialog->rtptimeout) {
+ while (dialog->owner && ast_channel_trylock(dialog->owner)) {
+ sip_pvt_unlock(dialog);
+ usleep(1);
+ sip_pvt_lock(dialog);
+ }
+ if (!(ast_rtp_get_bridged(dialog->rtp))) {
+ ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
+ dialog->owner->name, (long) (t - dialog->lastrtprx));
+ /* Issue a softhangup */
+ ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
+ } else
+ ast_log(LOG_NOTICE, "'%s' will not be disconnected in %ld seconds because it is directly bridged to another RTP stream\n", dialog->owner->name, (long) (t - dialog->lastrtprx));
+ ast_channel_unlock(dialog->owner);
+ /* forget the timeouts for this call, since a hangup
+ has already been requested and we don't want to
+ repeatedly request hangups
+ */
+ ast_rtp_set_rtptimeout(dialog->rtp, 0);
+ ast_rtp_set_rtpholdtimeout(dialog->rtp, 0);
+ if (dialog->vrtp) {
+ ast_rtp_set_rtptimeout(dialog->vrtp, 0);
+ ast_rtp_set_rtpholdtimeout(dialog->vrtp, 0);
}
}
}
@@ -15799,7 +15815,7 @@
static void *do_monitor(void *data)
{
int res;
- struct sip_pvt *sip;
+ struct sip_pvt *dialog;
struct sip_peer *peer = NULL;
time_t t;
int fastrestart = FALSE;
@@ -15827,6 +15843,7 @@
if (sipsock > -1)
sipsock_read_id = ast_io_change(io, sipsock_read_id, sipsock, NULL, 0, NULL);
}
+
/* Check for dialogs needing to be killed */
dialoglist_lock();
restartsearch:
@@ -15835,18 +15852,20 @@
of time since the last time we did it (when MWI is being sent, we can
get back to this point every millisecond or less)
*/
- for (sip = dialoglist; !fastrestart && sip; sip = sip->next) {
- sip_pvt_lock(sip);
+ for (dialog = dialoglist; !fastrestart && dialog; dialog = dialog->next) {
+ sip_pvt_lock(dialog);
/* Check RTP timeouts and kill calls if we have a timeout set and do not get RTP */
- check_rtp_timeout(sip, t);
+ check_rtp_timeout(dialog, t);
/* If we have sessions that needs to be destroyed, do it now */
- if (ast_test_flag(&sip->flags[0], SIP_NEEDDESTROY) && !sip->packets &&
- !sip->owner) {
- sip_pvt_unlock(sip);
- __sip_destroy(sip, TRUE, FALSE);
+ /* Check if we have outstanding requests not responsed to or an active call
+ - if that's the case, wait with destruction */
+ if (ast_test_flag(&dialog->flags[0], SIP_NEEDDESTROY) && !dialog->packets &&
+ !dialog->owner) {
+ sip_pvt_unlock(dialog);
+ __sip_destroy(dialog, TRUE, FALSE);
goto restartsearch;
}
- sip_pvt_unlock(sip);
+ sip_pvt_unlock(dialog);
}
dialoglist_unlock();
@@ -15855,7 +15874,8 @@
res = ast_sched_wait(sched);
if ((res < 0) || (res > 1000))
res = 1000;
- /* If we might need to send more mailboxes, don't wait long at all.*/
+
+ /* If we might need to send more mailbox notifications, don't wait long at all.*/
if (fastrestart)
res = 1;
res = ast_io_wait(io, res);
@@ -17144,10 +17164,10 @@
compactheaders = ast_true(v->value);
} else if (!strcasecmp(v->name, "notifymimetype")) {
ast_copy_string(default_notifymime, v->value, sizeof(default_notifymime));
+ } else if (!strcasecmp(v->name, "limitonpeers")) {
+ global_limitonpeers = ast_true(v->value);
} else if (!strcasecmp(v->name, "notifyringing")) {
global_notifyringing = ast_true(v->value);
- } else if (!strcasecmp(v->name, "limitpeersonly")) {
- global_limitonpeers = ast_true(v->value);
} else if (!strcasecmp(v->name, "notifyhold")) {
global_notifyhold = ast_true(v->value);
} else if (!strcasecmp(v->name, "alwaysauthreject")) {
@@ -17185,7 +17205,7 @@
autocreatepeer = ast_true(v->value);
} else if (!strcasecmp(v->name, "match_auth_username")) {
global_match_auth_username = ast_true(v->value);
- } else if (!strcasecmp(v->name, "global_srvlookup")) {
+ } else if (!strcasecmp(v->name, "srvlookup")) {
global_srvlookup = ast_true(v->value);
} else if (!strcasecmp(v->name, "pedantic")) {
pedanticsipchecking = ast_true(v->value);
@@ -18047,6 +18067,7 @@
sip_destroy(iterator->call);
}
ASTOBJ_UNLOCK(iterator);
+
} while(0));
/* Then, actually destroy users and registry */
Modified: team/oej/videocaps/channels/chan_zap.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/channels/chan_zap.c?view=diff&rev=48211&r1=48210&r2=48211
==============================================================================
--- team/oej/videocaps/channels/chan_zap.c (original)
+++ team/oej/videocaps/channels/chan_zap.c Sat Dec 2 12:54:41 2006
@@ -8380,7 +8380,7 @@
startcic = linkset->pvts[0]->cic;
for (i = 0; i < linkset->numchans; i++) {
- if (linkset->pvts[i+1] && (linkset->pvts[i+1]->cic - linkset->pvts[i]->cic) == 1) {
+ if (linkset->pvts[i+1] && ((linkset->pvts[i+1]->cic - linkset->pvts[i]->cic) == 1) && (linkset->pvts[i]->cic - startcic < 31)) {
continue;
} else {
endcic = linkset->pvts[i]->cic;
@@ -10568,23 +10568,23 @@
return RESULT_SUCCESS;
}
-static const char pri_debug_help[] =
+static char pri_debug_help[] =
"Usage: pri debug span <span>\n"
" Enables debugging on a given PRI span\n";
-static const char pri_no_debug_help[] =
+static char pri_no_debug_help[] =
"Usage: pri no debug span <span>\n"
" Disables debugging on a given PRI span\n";
-static const char pri_really_debug_help[] =
+static char pri_really_debug_help[] =
"Usage: pri intensive debug span <span>\n"
" Enables debugging down to the Q.921 level\n";
-static const char pri_show_span_help[] =
+static char pri_show_span_help[] =
"Usage: pri show span <span>\n"
" Displays PRI Information on a given PRI span\n";
-static const char pri_show_spans_help[] =
+static char pri_show_spans_help[] =
"Usage: pri show spans\n"
" Displays PRI Information\n";
@@ -11556,7 +11556,6 @@
return RESULT_SUCCESS;
}
-#if 0
static int handle_ss7_show_linkset(int fd, int argc, char *argv[])
{
int linkset;
@@ -11573,49 +11572,44 @@
return RESULT_SUCCESS;
}
if (linksets[linkset-1].ss7)
- ss7 = linksets[linkset-1];
-
- if (
+ ss7 = &linksets[linkset-1];
+
+ ast_cli(fd, "SS7 linkset %d status: %s\n", linkset, (ss7->state == LINKSET_STATE_UP) ? "Up" : "Down");
return RESULT_SUCCESS;
}
-#endif
-
-static const char ss7_debug_help[] =
+
+static char ss7_debug_help[] =
"Usage: ss7 debug linkset <linkset>\n"
" Enables debugging on a given SS7 linkset\n";
-static const char ss7_no_debug_help[] =
+static char ss7_no_debug_help[] =
"Usage: ss7 no debug linkset <span>\n"
" Disables debugging on a given SS7 linkset\n";
-static const char ss7_block_cic_help[] =
+static char ss7_block_cic_help[] =
"Usage: ss7 block cic <linkset> <CIC>\n"
" Sends a remote blocking request for the given CIC on the specified linkset\n";
-static const char ss7_unblock_cic_help[] =
+static char ss7_unblock_cic_help[] =
"Usage: ss7 unblock cic <linkset> <CIC>\n"
" Sends a remote unblocking request for the given CIC on the specified linkset\n";
-#if 0
-static const char ss7_show_linkset_help[] =
+static char ss7_show_linkset_help[] =
"Usage: ss7 show linkset <span>\n"
- " Disables debugging on a given SS7 linkset\n";
-#endif
+ " Shows the status of an SS7 linkset.\n";
static struct ast_cli_entry zap_ss7_cli[] = {
{ { "ss7", "debug", "linkset", NULL }, handle_ss7_debug,
"Enables SS7 debugging on a linkset", ss7_debug_help, NULL },
{ { "ss7", "no", "debug", "linkset", NULL }, handle_ss7_no_debug,
- "Disables SS7 debugging on a linkset", ss7_debug_help, NULL },
+ "Disables SS7 debugging on a linkset", ss7_no_debug_help, NULL },
{ { "ss7", "block", "cic", NULL }, handle_ss7_block_cic,
"Disables SS7 debugging on a linkset", ss7_block_cic_help, NULL },
{ { "ss7", "unblock", "cic", NULL }, handle_ss7_unblock_cic,
"Disables SS7 debugging on a linkset", ss7_unblock_cic_help, NULL },
-#if 0
{ { "ss7", "show", "linkset", NULL }, handle_ss7_show_linkset,
- "Disables SS7 debugging on a linkset", ss7_show_linkset_help, NULL },
-#endif
+ "Shows the status of a linkset", ss7_show_linkset_help, NULL },
};
#endif /* HAVE_SS7 */
Modified: team/oej/videocaps/codecs/codec_zap.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/codecs/codec_zap.c?view=diff&rev=48211&r1=48210&r2=48211
==============================================================================
--- team/oej/videocaps/codecs/codec_zap.c (original)
+++ team/oej/videocaps/codecs/codec_zap.c Sat Dec 2 12:54:41 2006
@@ -232,7 +232,7 @@
return zap_translate(pvt, pvt->t->dstfmt, pvt->t->srcfmt);
}
-static struct ast_frame *g729_fakesrc_sample()
+static struct ast_frame *g729_fakesrc_sample(void)
{
/* Don't bother really trying to test hardware ones. */
static struct ast_frame f = {
@@ -244,7 +244,7 @@
return &f;
}
-static struct ast_frame *g723_fakesrc_sample()
[... 3392 lines stripped ...]
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