[svn-commits] oej: branch oej/earlyrtpfix r48262 - in
/team/oej/earlyrtpfix: ./ apps/ chann...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Tue Dec 5 05:28:17 MST 2006
Author: oej
Date: Tue Dec 5 06:28:17 2006
New Revision: 48262
URL: http://svn.digium.com/view/asterisk?view=rev&rev=48262
Log:
Updating branch to trunk status
We might have to go a different path for 1.4 and disable the direct RTP
media by default until we have found a solution for all this.
Modified:
team/oej/earlyrtpfix/ (props changed)
team/oej/earlyrtpfix/Makefile
team/oej/earlyrtpfix/apps/app_dial.c
team/oej/earlyrtpfix/apps/app_voicemail.c
team/oej/earlyrtpfix/channels/chan_gtalk.c
team/oej/earlyrtpfix/channels/chan_iax2.c
team/oej/earlyrtpfix/channels/chan_phone.c
team/oej/earlyrtpfix/channels/chan_sip.c
team/oej/earlyrtpfix/configs/extensions.conf.sample
team/oej/earlyrtpfix/configs/sip.conf.sample
team/oej/earlyrtpfix/configs/voicemail.conf.sample
team/oej/earlyrtpfix/configure
team/oej/earlyrtpfix/configure.ac
team/oej/earlyrtpfix/doc/manager.txt
team/oej/earlyrtpfix/doc/snmp.txt
team/oej/earlyrtpfix/include/asterisk/rtp.h
team/oej/earlyrtpfix/include/asterisk/utils.h
team/oej/earlyrtpfix/main/cdr.c
team/oej/earlyrtpfix/main/cli.c
team/oej/earlyrtpfix/main/rtp.c
team/oej/earlyrtpfix/makeopts.in
team/oej/earlyrtpfix/res/res_features.c
team/oej/earlyrtpfix/sounds/Makefile
Propchange: team/oej/earlyrtpfix/
------------------------------------------------------------------------------
Binary property 'branch-1.2-blocked' - no diff available.
Propchange: team/oej/earlyrtpfix/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.
Propchange: team/oej/earlyrtpfix/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Dec 5 06:28:17 2006
@@ -1,1 +1,1 @@
-/branches/1.4:1-48071
+/branches/1.4:1-48261
Modified: team/oej/earlyrtpfix/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/Makefile?view=diff&rev=48262&r1=48261&r2=48262
==============================================================================
--- team/oej/earlyrtpfix/Makefile (original)
+++ team/oej/earlyrtpfix/Makefile Tue Dec 5 06:28:17 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/earlyrtpfix/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/apps/app_dial.c?view=diff&rev=48262&r1=48261&r2=48262
==============================================================================
--- team/oej/earlyrtpfix/apps/app_dial.c (original)
+++ team/oej/earlyrtpfix/apps/app_dial.c Tue Dec 5 06:28:17 2006
@@ -1233,6 +1233,7 @@
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/earlyrtpfix/apps/app_voicemail.c
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/apps/app_voicemail.c?view=diff&rev=48262&r1=48261&r2=48262
==============================================================================
--- team/oej/earlyrtpfix/apps/app_voicemail.c (original)
+++ team/oej/earlyrtpfix/apps/app_voicemail.c Tue Dec 5 06:28:17 2006
@@ -142,6 +142,7 @@
/* Don't modify these here; set your umask at runtime instead */
#define VOICEMAIL_DIR_MODE 0777
#define VOICEMAIL_FILE_MODE 0666
+#define CHUNKSIZE 65536
#define VOICEMAIL_CONFIG "voicemail.conf"
#define ASTERISK_USERNAME "asterisk"
@@ -1088,6 +1089,7 @@
goto yuck;
}
if (!strcasecmp(coltitle, "recording")) {
+ off_t offset;
res = SQLGetData(stmt, x + 1, SQL_BINARY, NULL, 0, &colsize2);
fdlen = colsize2;
if (fd > -1) {
@@ -1098,24 +1100,27 @@
fd = -1;
continue;
}
- if (fd > -1) {
- if ((fdm = mmap(NULL, fdlen, PROT_READ | PROT_WRITE, MAP_SHARED, fd, 0)) == -1) {
+ /* Read out in small chunks */
+ for (offset = 0; offset < colsize2; offset += CHUNKSIZE) {
+ /* +1 because SQLGetData likes null-terminating binary data */
+ if ((fdm = mmap(NULL, CHUNKSIZE + 1, PROT_READ | PROT_WRITE, MAP_SHARED, fd, offset)) == (void *)-1) {
ast_log(LOG_WARNING, "Could not mmap the output file: %s (%d)\n", strerror(errno), errno);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
ast_odbc_release_obj(obj);
goto yuck;
+ } else {
+ res = SQLGetData(stmt, x + 1, SQL_BINARY, fdm, CHUNKSIZE + 1, NULL);
+ munmap(fdm, 0);
+ if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
+ ast_log(LOG_WARNING, "SQL Get Data error!\n[%s]\n\n", sql);
+ unlink(full_fn);
+ SQLFreeHandle(SQL_HANDLE_STMT, stmt);
+ ast_odbc_release_obj(obj);
+ goto yuck;
+ }
}
}
- }
- if (fdm) {
- memset(fdm, 0, fdlen);
- res = SQLGetData(stmt, x + 1, SQL_BINARY, fdm, fdlen, &colsize2);
- if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
- ast_log(LOG_WARNING, "SQL Get Data error!\n[%s]\n\n", sql);
- SQLFreeHandle (SQL_HANDLE_STMT, stmt);
- ast_odbc_release_obj(obj);
- goto yuck;
- }
+ truncate(full_fn, fdlen);
}
} else {
res = SQLGetData(stmt, x + 1, SQL_CHAR, rowdata, sizeof(rowdata), NULL);
@@ -1136,8 +1141,6 @@
yuck:
if (f)
fclose(f);
- if (fdm)
- munmap(fdm, fdlen);
if (fd > -1)
close(fd);
return x - 1;
@@ -4648,7 +4651,7 @@
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;
}
@@ -7876,6 +7879,8 @@
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Saving message as is\n");
ast_stream_and_wait(chan, "vm-msgsaved", chan->language, "");
+ STORE(recordfile, vmu->mailbox, vmu->context, -1, chan, vmu, fmt, duration, vms);
+ DISPOSE(recordfile, -1);
cmd = 't';
return res;
}
@@ -8345,15 +8350,16 @@
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/earlyrtpfix/channels/chan_gtalk.c
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/channels/chan_gtalk.c?view=diff&rev=48262&r1=48261&r2=48262
==============================================================================
--- team/oej/earlyrtpfix/channels/chan_gtalk.c (original)
+++ team/oej/earlyrtpfix/channels/chan_gtalk.c Tue Dec 5 06:28:17 2006
@@ -163,7 +163,6 @@
};
static const char desc[] = "Gtalk Channel";
-static const char type[] = "Gtalk";
static int usecnt = 0;
AST_MUTEX_DEFINE_STATIC(usecnt_lock);
@@ -195,7 +194,7 @@
/*! \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,
@@ -223,7 +222,7 @@
/*! \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,
@@ -922,10 +921,12 @@
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);
}
@@ -1796,7 +1797,7 @@
/* 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/earlyrtpfix/channels/chan_iax2.c
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/channels/chan_iax2.c?view=diff&rev=48262&r1=48261&r2=48262
==============================================================================
--- team/oej/earlyrtpfix/channels/chan_iax2.c (original)
+++ team/oej/earlyrtpfix/channels/chan_iax2.c Tue Dec 5 06:28:17 2006
@@ -6935,7 +6935,7 @@
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))
ast_log(LOG_DEBUG, "Parked call on '%s'\n", ast_bridged_channel(iaxs[fr->callno]->owner)->name);
} else {
if (ast_async_goto(ast_bridged_channel(iaxs[fr->callno]->owner), iaxs[fr->callno]->context, ies.called_number, 1))
Modified: team/oej/earlyrtpfix/channels/chan_phone.c
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/channels/chan_phone.c?view=diff&rev=48262&r1=48261&r2=48262
==============================================================================
--- team/oej/earlyrtpfix/channels/chan_phone.c (original)
+++ team/oej/earlyrtpfix/channels/chan_phone.c Tue Dec 5 06:28:17 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/earlyrtpfix/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/channels/chan_sip.c?view=diff&rev=48262&r1=48261&r2=48262
==============================================================================
--- team/oej/earlyrtpfix/channels/chan_sip.c (original)
+++ team/oej/earlyrtpfix/channels/chan_sip.c Tue Dec 5 06:28:17 2006
@@ -513,6 +513,7 @@
static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
/* Global settings only apply to the channel */
+static int global_limitonpeers; /*!< Match call limit on peers only */
static int global_rtautoclear;
static int global_notifyringing; /*!< Send notifications on ringing */
static int global_notifyhold; /*!< Send notifications on hold */
@@ -947,8 +948,6 @@
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) */
@@ -2596,17 +2595,21 @@
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);
- }
-
- /* Set Frame packetization */
- if (dialog->rtp) {
- ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
- dialog->autoframing = peer->autoframing;
- }
+ ast_rtp_set_rtptimeout(dialog->vrtp, peer->rtptimeout);
+ ast_rtp_set_rtpholdtimeout(dialog->vrtp, peer->rtpholdtimeout);
+ ast_rtp_set_rtpkeepalive(dialog->vrtp, peer->rtpkeepalive);
+ }
+
ast_string_field_set(dialog, peername, peer->username);
ast_string_field_set(dialog, authname, peer->username);
ast_string_field_set(dialog, username, peer->username);
@@ -2645,8 +2648,6 @@
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);
dialog->maxcallbitrate = peer->maxcallbitrate;
@@ -2948,7 +2949,7 @@
static int update_call_counter(struct sip_pvt *fup, int event)
{
char name[256];
- int *inuse, *call_limit, *inringing;
+ int *inuse = NULL, *call_limit = NULL, *inringing = NULL;
int outgoing = ast_test_flag(&fup->flags[0], SIP_OUTGOING);
struct sip_user *u = NULL;
struct sip_peer *p = NULL;
@@ -2963,16 +2964,17 @@
ast_copy_string(name, fup->username, sizeof(name));
/* Check the list of users only for incoming calls */
- if (!outgoing && (u = find_user(name, 1)) ) {
+ if (global_limitonpeers == FALSE && !outgoing && (u = find_user(name, 1))) {
inuse = &u->inUse;
call_limit = &u->call_limit;
inringing = NULL;
- } else if ( (p = find_peer(fup->peername, NULL, 1) ) ) { /* Try to find peer */
+ } 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;
inringing = &p->inRinging;
ast_copy_string(name, fup->peername, sizeof(name));
- } else {
+ }
+ if (!p && !u) {
if (option_debug > 1)
ast_log(LOG_DEBUG, "%s is not a local device, no call limit\n", name);
return 0;
@@ -3484,15 +3486,12 @@
case AST_FRAME_MODEM:
if (p) {
ast_mutex_lock(&p->lock);
- 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);
- }
ast_mutex_unlock(&p->lock);
}
break;
@@ -4187,16 +4186,19 @@
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->udptl)
ast_udptl_settos(p->udptl, global_tos_audio);
- p->rtptimeout = global_rtptimeout;
- p->rtpholdtimeout = global_rtpholdtimeout;
- p->rtpkeepalive = global_rtpkeepalive;
p->maxcallbitrate = default_maxcallbitrate;
}
@@ -7255,7 +7257,8 @@
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)))
@@ -10181,6 +10184,7 @@
ast_cli(fd, " Our auth realm %s\n", global_realm);
ast_cli(fd, " Realm. auth: %s\n", authl ? "Yes": "No");
ast_cli(fd, " Always auth rejects: %s\n", global_alwaysauthreject ? "Yes" : "No");
+ ast_cli(fd, " Call limit peers only: %s\n", global_limitonpeers ? "Yes" : "No");
ast_cli(fd, " User Agent: %s\n", global_useragent);
ast_cli(fd, " MWI checking interval: %d secs\n", global_mwitime);
ast_cli(fd, " Reg. context: %s\n", S_OR(global_regcontext, "(not set)"));
@@ -10210,6 +10214,7 @@
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);
@@ -11624,6 +11629,9 @@
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 */
@@ -11716,9 +11724,12 @@
break;
case 481: /* Call leg does not exist */
- /* Could be REFER or INVITE */
+ /* Could be REFER caused INVITE with replaces */
ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
+ if (p->owner)
+ ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
break;
case 491: /* Pending */
@@ -11778,7 +11789,16 @@
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
}
break;
-
+ case 481: /* Call leg does not exist */
+
+ /* A transfer with Replaces did not work */
+ /* OEJ: We should Set flag, cancel the REFER, go back
+ to original call - but right now we can't */
+ ast_log(LOG_WARNING, "Remote host can't match REFER request to call '%s'. Giving up.\n", p->callid);
+ if (p->owner)
+ ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+ ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ break;
case 500: /* Server error */
case 501: /* Method not implemented */
@@ -12113,21 +12133,9 @@
break;
case 481: /* Call leg does not exist */
if (sipmethod == SIP_INVITE) {
- /* First we ACK */
- transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- 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);
+ handle_response_invite(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_REFER) {
- /* A transfer with Replaces did not work */
- /* OEJ: We should Set flag, cancel the REFER, go back
- to original call - but right now we can't */
- ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid);
- if (owner)
- ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ handle_response_refer(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_BYE) {
/* The other side has no transaction to bye,
just assume it's all right then */
@@ -14350,7 +14358,7 @@
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;
}
@@ -14740,23 +14748,23 @@
(sip->owner->_state == AST_STATE_UP) &&
!sip->redirip.sin_addr.s_addr) {
if (sip->lastrtptx &&
- sip->rtpkeepalive &&
- (t > sip->lastrtptx + sip->rtpkeepalive)) {
+ ast_rtp_get_rtpkeepalive(sip->rtp) &&
+ (t > sip->lastrtptx + ast_rtp_get_rtpkeepalive(sip->rtp))) {
/* 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)) {
+ (ast_rtp_get_rtptimeout(sip->rtp) || ast_rtp_get_rtpholdtimeout(sip->rtp)) &&
+ (t > sip->lastrtprx + ast_rtp_get_rtptimeout(sip->rtp))) {
/* 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))) {
+ (ast_rtp_get_rtpholdtimeout(sip->rtp) &&
+ (t > sip->lastrtprx + ast_rtp_get_rtpholdtimeout(sip->rtp)))) {
/* Needs a hangup */
- if (sip->rtptimeout) {
+ if (ast_rtp_get_rtptimeout(sip->rtp)) {
while (sip->owner && ast_channel_trylock(sip->owner)) {
ast_mutex_unlock(&sip->lock);
usleep(1);
@@ -14777,8 +14785,12 @@
has already been requested and we don't want to
repeatedly request hangups
*/
- sip->rtptimeout = 0;
- sip->rtpholdtimeout = 0;
+ ast_rtp_set_rtptimeout(sip->rtp, 0);
+ ast_rtp_set_rtpholdtimeout(sip->rtp, 0);
+ if (sip->vrtp) {
+ ast_rtp_set_rtptimeout(sip->vrtp, 0);
+ ast_rtp_set_rtpholdtimeout(sip->vrtp, 0);
+ }
}
}
}
@@ -15918,6 +15930,7 @@
global_regcontext[0] = '\0';
expiry = DEFAULT_EXPIRY;
global_notifyringing = DEFAULT_NOTIFYRINGING;
+ global_limitonpeers = FALSE;
global_notifyhold = FALSE;
global_alwaysauthreject = 0;
global_allowsubscribe = FALSE;
@@ -16040,6 +16053,8 @@
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, "notifyhold")) {
Modified: team/oej/earlyrtpfix/configs/extensions.conf.sample
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/configs/extensions.conf.sample?view=diff&rev=48262&r1=48261&r2=48262
==============================================================================
--- team/oej/earlyrtpfix/configs/extensions.conf.sample (original)
+++ team/oej/earlyrtpfix/configs/extensions.conf.sample Tue Dec 5 06:28:17 2006
@@ -167,7 +167,7 @@
;
; List canonical entries here
;
-;exten => 12564286000,1,Macro(std-exten,6000,IAX2/foo)
+;exten => 12564286000,1,Macro(stdexten,6000,IAX2/foo)
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})
[dundi-e164-customers]
Modified: team/oej/earlyrtpfix/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/configs/sip.conf.sample?view=diff&rev=48262&r1=48261&r2=48262
==============================================================================
--- team/oej/earlyrtpfix/configs/sip.conf.sample (original)
+++ team/oej/earlyrtpfix/configs/sip.conf.sample Tue Dec 5 06:28:17 2006
@@ -35,6 +35,7 @@
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
+ ; bindport is the local UDP port that Asterisk will listen on
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
@@ -90,10 +91,6 @@
;language=en ; Default language setting for all users/peers
; This may also be set for individual users/peers
;relaxdtmf=yes ; Relax dtmf handling
-;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
- ; when we're not on hold
-;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity
- ; when we're on hold (must be > rtptimeout)
;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = yes ; If Remote-Party-ID should be sent
;progressinband=never ; If we should generate in-band ringing always
@@ -149,6 +146,21 @@
;
;regcontext=sipregistrations
;
+;--------------------------- RTP timers ----------------------------------------------------
+; These timers are currently used for both audio and video streams. The RTP timeouts
+; are only applied to the audio channel.
+; The settings are settable in the global section as well as per device
+;
+;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
+ ; on the audio channel
+ ; when we're not on hold. This is to be able to hangup
+ ; a call in the case of a phone disappearing from the net,
+ ; like a powerloss or grandma tripping over a cable.
+;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
+ ; on the audio channel
+ ; when we're on hold (must be > rtptimeout)
+;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
+ ; (default is off - zero)
;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration
@@ -162,6 +174,15 @@
; You can subscribe to the status of extensions with a "hint" priority
; (See extensions.conf.sample for examples)
; chan_sip support two major formats for notifications: dialog-info and SIMPLE
+;
+; You will get more detailed reports (busy etc) if you have a call limit set
+; for a device. When the call limit is filled, we will indicate busy. Note that
+; you need at least 2 in order to be able to do attended transfers.
+;
+; For queues, you will need this level of detail in status reporting, regardless
+; if you use SIP subscriptions. Queues and manager use the same internal interface
+; for reading status information.
+;
; Note: Subscriptions does not work if you have a realtime dialplan and use the
; realtime switch.
;
@@ -173,13 +194,20 @@
;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
; Turning on notifyringing and notifyhold will add a lot
; more database transactions if you are using realtime.
+;limitonpeer = yes ; Apply call limits on peers only. This will improve
+ ; status notification when you are using type=friend
+ ; Inbound calls, that really apply to the user part
+ ; of a friend will now be added to and compared with
+ ; the peer limit instead of applying two call limits,
+ ; one for the peer and one for the user.
;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
;
; This setting is available in the [general] section as well as in device configurations.
; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
-; both parties have T38 support enabled in their Asterisk configuration (either general or
-; peer/user/friend sections)
+; both parties have T38 support enabled in their Asterisk configuration
+; This has to be enabled in the general section for all devices to work. You can then
+; disable it on a per device basis.
;
; t38pt_udptl = yes ; Default false
;
@@ -470,8 +498,9 @@
;usereqphone=yes ; This provider requires ";user=phone" on URI
;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
- ; Call-limits will not be enforced on real-time peers,
- ; since they are not stored in-memory
+ ; Call-limits will not be enforced on real-time peers,
+ ; since they are not stored in-memory
+;port=80 ; The port number we want to connect to on the remote side
;------------------------------------------------------------------------------
; Definitions of locally connected SIP devices
Modified: team/oej/earlyrtpfix/configs/voicemail.conf.sample
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/configs/voicemail.conf.sample?view=diff&rev=48262&r1=48261&r2=48262
==============================================================================
--- team/oej/earlyrtpfix/configs/voicemail.conf.sample (original)
+++ team/oej/earlyrtpfix/configs/voicemail.conf.sample Tue Dec 5 06:28:17 2006
@@ -83,6 +83,12 @@
;fromstring=The Asterisk PBX
; Permit finding entries for forward/compose from the directory
;usedirectory=yes
+; Voicemail can be stored in a database using the ODBC driver.
+; The value of odbcstorage is the database connection configured
+; in res_odbc.conf.
+;odbcstorage=asterisk
+; The default table for ODBC voicemail storage is voicemessages.
+;odbctable=voicemessages
;
; Change the from, body and/or subject, variables:
; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM,
Modified: team/oej/earlyrtpfix/configure
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/configure?view=diff&rev=48262&r1=48261&r2=48262
==============================================================================
--- team/oej/earlyrtpfix/configure (original)
+++ team/oej/earlyrtpfix/configure Tue Dec 5 06:28:17 2006
@@ -1,5 +1,5 @@
#! /bin/sh
-# From configure.ac Revision: 47327 .
+# From configure.ac Revision: 47758 .
# Guess values for system-dependent variables and create Makefiles.
# Generated by GNU Autoconf 2.60a.
#
@@ -691,6 +691,9 @@
LN
DOT
STRIP
+WGET
+FETCH
+DOWNLOAD
AST_DEVMODE
ALSA_LIB
ALSA_INCLUDE
@@ -5568,6 +5571,94 @@
fi
+# Extract the first word of "wget", so it can be a program name with args.
+set dummy wget; ac_word=$2
+{ echo "$as_me:$LINENO: checking for $ac_word" >&5
+echo $ECHO_N "checking for $ac_word... $ECHO_C" >&6; }
+if test "${ac_cv_path_WGET+set}" = set; then
+ echo $ECHO_N "(cached) $ECHO_C" >&6
+else
+ case $WGET in
+ [\\/]* | ?:[\\/]*)
+ ac_cv_path_WGET="$WGET" # Let the user override the test with a path.
+ ;;
+ *)
+ as_save_IFS=$IFS; IFS=$PATH_SEPARATOR
+for as_dir in $PATH
+do
+ IFS=$as_save_IFS
+ test -z "$as_dir" && as_dir=.
+ for ac_exec_ext in '' $ac_executable_extensions; do
+ if { test -f "$as_dir/$ac_word$ac_exec_ext" && $as_executable_p "$as_dir/$ac_word$ac_exec_ext"; }; then
+ ac_cv_path_WGET="$as_dir/$ac_word$ac_exec_ext"
+ echo "$as_me:$LINENO: found $as_dir/$ac_word$ac_exec_ext" >&5
+ break 2
+ fi
+done
+done
+IFS=$as_save_IFS
+
+ test -z "$ac_cv_path_WGET" && ac_cv_path_WGET=":"
+ ;;
+esac
+fi
+WGET=$ac_cv_path_WGET
+if test -n "$WGET"; then
+ { echo "$as_me:$LINENO: result: $WGET" >&5
+echo "${ECHO_T}$WGET" >&6; }
+else
+ { echo "$as_me:$LINENO: result: no" >&5
+echo "${ECHO_T}no" >&6; }
+fi
+
+
+if test "${WGET}" != ":" ; then
+ DOWNLOAD=${WGET}
+else
+ # Extract the first word of "fetch", so it can be a program name with args.
+set dummy fetch; ac_word=$2
+{ echo "$as_me:$LINENO: checking for $ac_word" >&5
+echo $ECHO_N "checking for $ac_word... $ECHO_C" >&6; }
+if test "${ac_cv_path_FETCH+set}" = set; then
+ echo $ECHO_N "(cached) $ECHO_C" >&6
+else
+ case $FETCH in
+ [\\/]* | ?:[\\/]*)
+ ac_cv_path_FETCH="$FETCH" # Let the user override the test with a path.
+ ;;
+ *)
+ as_save_IFS=$IFS; IFS=$PATH_SEPARATOR
+for as_dir in $PATH
+do
+ IFS=$as_save_IFS
+ test -z "$as_dir" && as_dir=.
+ for ac_exec_ext in '' $ac_executable_extensions; do
+ if { test -f "$as_dir/$ac_word$ac_exec_ext" && $as_executable_p "$as_dir/$ac_word$ac_exec_ext"; }; then
+ ac_cv_path_FETCH="$as_dir/$ac_word$ac_exec_ext"
+ echo "$as_me:$LINENO: found $as_dir/$ac_word$ac_exec_ext" >&5
+ break 2
+ fi
+done
+done
+IFS=$as_save_IFS
+
+ test -z "$ac_cv_path_FETCH" && ac_cv_path_FETCH=":"
+ ;;
+esac
+fi
+FETCH=$ac_cv_path_FETCH
+if test -n "$FETCH"; then
+ { echo "$as_me:$LINENO: result: $FETCH" >&5
+echo "${ECHO_T}$FETCH" >&6; }
+else
+ { echo "$as_me:$LINENO: result: no" >&5
+echo "${ECHO_T}no" >&6; }
+fi
+
+
+ DOWNLOAD=${FETCH}
+fi
+
ac_ext=c
ac_cpp='$CPP $CPPFLAGS'
@@ -31677,14 +31768,14 @@
LN!$LN$ac_delim
DOT!$DOT$ac_delim
STRIP!$STRIP$ac_delim
+WGET!$WGET$ac_delim
+FETCH!$FETCH$ac_delim
+DOWNLOAD!$DOWNLOAD$ac_delim
AST_DEVMODE!$AST_DEVMODE$ac_delim
ALSA_LIB!$ALSA_LIB$ac_delim
ALSA_INCLUDE!$ALSA_INCLUDE$ac_delim
PBX_ALSA!$PBX_ALSA$ac_delim
CURL_LIB!$CURL_LIB$ac_delim
-CURL_INCLUDE!$CURL_INCLUDE$ac_delim
-PBX_CURL!$PBX_CURL$ac_delim
-CURSES_LIB!$CURSES_LIB$ac_delim
_ACEOF
if test `sed -n "s/.*$ac_delim\$/X/p" conf$$subs.sed | grep -c X` = 97; then
@@ -31726,6 +31817,9 @@
ac_delim='%!_!# '
for ac_last_try in false false false false false :; do
cat >conf$$subs.sed <<_ACEOF
+CURL_INCLUDE!$CURL_INCLUDE$ac_delim
+PBX_CURL!$PBX_CURL$ac_delim
+CURSES_LIB!$CURSES_LIB$ac_delim
CURSES_INCLUDE!$CURSES_INCLUDE$ac_delim
PBX_CURSES!$PBX_CURSES$ac_delim
GNUTLS_LIB!$GNUTLS_LIB$ac_delim
@@ -31820,9 +31914,6 @@
PBX_TONEZONE!$PBX_TONEZONE$ac_delim
VORBIS_LIB!$VORBIS_LIB$ac_delim
VORBIS_INCLUDE!$VORBIS_INCLUDE$ac_delim
-PBX_VORBIS!$PBX_VORBIS$ac_delim
-VPB_LIB!$VPB_LIB$ac_delim
-VPB_INCLUDE!$VPB_INCLUDE$ac_delim
_ACEOF
if test `sed -n "s/.*$ac_delim\$/X/p" conf$$subs.sed | grep -c X` = 97; then
@@ -31864,6 +31955,9 @@
ac_delim='%!_!# '
for ac_last_try in false false false false false :; do
cat >conf$$subs.sed <<_ACEOF
+PBX_VORBIS!$PBX_VORBIS$ac_delim
+VPB_LIB!$VPB_LIB$ac_delim
+VPB_INCLUDE!$VPB_INCLUDE$ac_delim
PBX_VPB!$PBX_VPB$ac_delim
ZLIB_LIB!$ZLIB_LIB$ac_delim
ZLIB_INCLUDE!$ZLIB_INCLUDE$ac_delim
@@ -31903,7 +31997,7 @@
LTLIBOBJS!$LTLIBOBJS$ac_delim
_ACEOF
- if test `sed -n "s/.*$ac_delim\$/X/p" conf$$subs.sed | grep -c X` = 37; then
+ if test `sed -n "s/.*$ac_delim\$/X/p" conf$$subs.sed | grep -c X` = 40; then
break
elif $ac_last_try; then
{ { echo "$as_me:$LINENO: error: could not make $CONFIG_STATUS" >&5
Modified: team/oej/earlyrtpfix/configure.ac
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/configure.ac?view=diff&rev=48262&r1=48261&r2=48262
==============================================================================
--- team/oej/earlyrtpfix/configure.ac (original)
+++ team/oej/earlyrtpfix/configure.ac Tue Dec 5 06:28:17 2006
@@ -149,6 +149,14 @@
AC_PATH_PROG([LN], [ln], :)
AC_PATH_PROG([DOT], [dot], :)
AC_PATH_PROG([STRIP], [strip], :)
+AC_PATH_PROG([WGET], [wget], :)
+if test "${WGET}" != ":" ; then
+ DOWNLOAD=${WGET}
+else
+ AC_PATH_PROG([FETCH], [fetch], [:])
+ DOWNLOAD=${FETCH}
+fi
+AC_SUBST(DOWNLOAD)
AC_LANG(C)
Modified: team/oej/earlyrtpfix/doc/manager.txt
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/doc/manager.txt?view=diff&rev=48262&r1=48261&r2=48262
==============================================================================
--- team/oej/earlyrtpfix/doc/manager.txt (original)
+++ team/oej/earlyrtpfix/doc/manager.txt Tue Dec 5 06:28:17 2006
@@ -30,6 +30,15 @@
If you develop applications, please try to reuse existing manager
headers and their interpretation. If you are unsure, discuss on
the asterisk-dev mailing list.
+
+Device status reports
+---------------------
+Manager subscribes to extension status reports from all channels,
+to be able to generate events when an extension or device changes
+state. The level of details in these events may depend on the channel
+and device configuration. Please check each channel configuration
+file for more information. (in sip.conf, check the section on
+subscriptions and call limits)
Command Syntax
Modified: team/oej/earlyrtpfix/doc/snmp.txt
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/doc/snmp.txt?view=diff&rev=48262&r1=48261&r2=48262
==============================================================================
--- team/oej/earlyrtpfix/doc/snmp.txt (original)
+++ team/oej/earlyrtpfix/doc/snmp.txt Tue Dec 5 06:28:17 2006
@@ -8,8 +8,11 @@
Note that on some (many?) Linux-distributions the dependency list in
the net-snmp-devel list is not complete, and additional RPMs will need
to be installed. This is typically seen as attempts to build res_snmp
-as net-snmp-devel is available, but then failures to find certain
-libraries.
+as net-snmp-devel is available, but then fails to find certain
+libraries. The packages may include the following:
+ * bzip2-devel
+ * lm_sensors-devel
+ * newt-devel
SNMP support comes in two varieties -- as a sub-agent to a running SNMP
daemon using the AgentX protocol, or as a full standalone agent. If
Modified: team/oej/earlyrtpfix/include/asterisk/rtp.h
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/include/asterisk/rtp.h?view=diff&rev=48262&r1=48261&r2=48262
==============================================================================
--- team/oej/earlyrtpfix/include/asterisk/rtp.h (original)
+++ team/oej/earlyrtpfix/include/asterisk/rtp.h Tue Dec 5 06:28:17 2006
@@ -186,6 +186,9 @@
/*! \brief Compensate for devices that send RFC2833 packets all at once */
void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate);
+/*! \brief Enable STUN capability */
+void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable);
+
int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
@@ -216,6 +219,21 @@
int ast_rtp_codec_getformat(int pt);
+/*! \brief Set rtp timeout */
+void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout);
+/*! \brief Set rtp hold timeout */
+void ast_rtp_set_rtpholdtimeout(struct ast_rtp *rtp, int timeout);
+/*! \brief set RTP keepalive interval */
+void ast_rtp_set_rtpkeepalive(struct ast_rtp *rtp, int period);
+/*! \brief Get RTP keepalive interval */
+int ast_rtp_get_rtpkeepalive(struct ast_rtp *rtp);
+/*! \brief Get rtp hold timeout */
+int ast_rtp_get_rtpholdtimeout(struct ast_rtp *rtp);
+/*! \brief Get rtp timeout */
+int ast_rtp_get_rtptimeout(struct ast_rtp *rtp);
[... 360 lines stripped ...]
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