[asterisk-commits] oej: branch oej/midcomstuff r48310 - in
/team/oej/midcomstuff: ./ apps/ chann...
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Wed Dec 6 01:20:55 MST 2006
Author: oej
Date: Wed Dec 6 02:20:55 2006
New Revision: 48310
URL: http://svn.digium.com/view/asterisk?view=rev&rev=48310
Log:
This is Asterisk-1.4-netsec, now updated to 1.4 svn
Modified:
team/oej/midcomstuff/ (props changed)
team/oej/midcomstuff/Makefile
team/oej/midcomstuff/apps/app_dial.c
team/oej/midcomstuff/apps/app_voicemail.c
team/oej/midcomstuff/channels/chan_gtalk.c
team/oej/midcomstuff/channels/chan_iax2.c
team/oej/midcomstuff/channels/chan_phone.c
team/oej/midcomstuff/channels/chan_sip.c
team/oej/midcomstuff/configs/extensions.conf.sample
team/oej/midcomstuff/configs/sip.conf.sample
team/oej/midcomstuff/configs/voicemail.conf.sample
team/oej/midcomstuff/configure
team/oej/midcomstuff/configure.ac
team/oej/midcomstuff/doc/manager.txt
team/oej/midcomstuff/doc/snmp.txt
team/oej/midcomstuff/include/asterisk/rtp.h
team/oej/midcomstuff/include/asterisk/utils.h
team/oej/midcomstuff/main/cdr.c
team/oej/midcomstuff/main/cli.c
team/oej/midcomstuff/main/manager.c
team/oej/midcomstuff/main/rtp.c
team/oej/midcomstuff/makeopts.in
team/oej/midcomstuff/pbx/pbx_spool.c
team/oej/midcomstuff/res/res_features.c
team/oej/midcomstuff/res/res_musiconhold.c
team/oej/midcomstuff/sounds/Makefile
Propchange: team/oej/midcomstuff/
------------------------------------------------------------------------------
Binary property 'branch-1.2-blocked' - no diff available.
Propchange: team/oej/midcomstuff/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.
Propchange: team/oej/midcomstuff/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Wed Dec 6 02:20:55 2006
@@ -1,1 +1,1 @@
-/branches/1.4:1-48025
+/branches/1.4:1-48307
Modified: team/oej/midcomstuff/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/midcomstuff/Makefile?view=diff&rev=48310&r1=48309&r2=48310
==============================================================================
--- team/oej/midcomstuff/Makefile (original)
+++ team/oej/midcomstuff/Makefile Wed Dec 6 02:20:55 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/midcomstuff/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/oej/midcomstuff/apps/app_dial.c?view=diff&rev=48310&r1=48309&r2=48310
==============================================================================
--- team/oej/midcomstuff/apps/app_dial.c (original)
+++ team/oej/midcomstuff/apps/app_dial.c Wed Dec 6 02:20:55 2006
@@ -1231,6 +1231,7 @@
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/midcomstuff/apps/app_voicemail.c
URL: http://svn.digium.com/view/asterisk/team/oej/midcomstuff/apps/app_voicemail.c?view=diff&rev=48310&r1=48309&r2=48310
==============================================================================
--- team/oej/midcomstuff/apps/app_voicemail.c (original)
+++ team/oej/midcomstuff/apps/app_voicemail.c Wed Dec 6 02:20:55 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;
}
@@ -6875,22 +6878,13 @@
}
}
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);
}
}
@@ -7885,6 +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;
}
@@ -8354,15 +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/midcomstuff/channels/chan_gtalk.c
URL: http://svn.digium.com/view/asterisk/team/oej/midcomstuff/channels/chan_gtalk.c?view=diff&rev=48310&r1=48309&r2=48310
==============================================================================
--- team/oej/midcomstuff/channels/chan_gtalk.c (original)
+++ team/oej/midcomstuff/channels/chan_gtalk.c Wed Dec 6 02:20:55 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/midcomstuff/channels/chan_iax2.c
URL: http://svn.digium.com/view/asterisk/team/oej/midcomstuff/channels/chan_iax2.c?view=diff&rev=48310&r1=48309&r2=48310
==============================================================================
--- team/oej/midcomstuff/channels/chan_iax2.c (original)
+++ team/oej/midcomstuff/channels/chan_iax2.c Wed Dec 6 02:20:55 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/midcomstuff/channels/chan_phone.c
URL: http://svn.digium.com/view/asterisk/team/oej/midcomstuff/channels/chan_phone.c?view=diff&rev=48310&r1=48309&r2=48310
==============================================================================
--- team/oej/midcomstuff/channels/chan_phone.c (original)
+++ team/oej/midcomstuff/channels/chan_phone.c Wed Dec 6 02:20:55 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/midcomstuff/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/midcomstuff/channels/chan_sip.c?view=diff&rev=48310&r1=48309&r2=48310
==============================================================================
--- team/oej/midcomstuff/channels/chan_sip.c (original)
+++ team/oej/midcomstuff/channels/chan_sip.c Wed Dec 6 02:20:55 2006
@@ -249,6 +249,21 @@
AST_FAILURE = -1,
};
+/*! \brief States for the INVITE transaction, not the dialog
+ \note this is for the INVITE that sets up the dialog
+*/
+enum invitestates {
+ INV_NONE = 0, /*!< No state at all, maybe not an INVITE dialog */
+ INV_CALLING = 1, /*!< Invite sent, no answer */
+ INV_PROCEEDING = 2, /*!< We got/sent 1xx message */
+ INV_EARLY_MEDIA = 3, /*!< We got 18x message with to-tag back */
+ INV_COMPLETED = 4, /*!< Got final response with error. Wait for ACK, then CONFIRMED */
+ INV_CONFIRMED = 5, /*!< Confirmed response - we've got an ack (Incoming calls only) */
+ INV_TERMINATED = 6, /*!< Transaction done - either successful (AST_STATE_UP) or failed, but done
+ The only way out of this is a BYE from one side */
+ INV_CANCELLED = 7, /*!< Transaction cancelled by client or server in non-terminated state */
+};
+
/* Do _NOT_ make any changes to this enum, or the array following it;
if you think you are doing the right thing, you are probably
not doing the right thing. If you think there are changes
@@ -518,6 +533,7 @@
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 */
@@ -707,7 +723,7 @@
#define SIP_REALTIME (1 << 11) /*!< Flag for realtime users */
#define SIP_USECLIENTCODE (1 << 12) /*!< Trust X-ClientCode info message */
#define SIP_OUTGOING (1 << 13) /*!< Direction of the last transaction in this dialog */
-#define SIP_CAN_BYE (1 << 14) /*!< Can we send BYE on this dialog? */
+#define SIP_FREE_BIT (1 << 14) /*!< ---- */
#define SIP_DEFER_BYE_ON_TRANSFER (1 << 15) /*!< Do not hangup at first ast_hangup */
#define SIP_DTMF (3 << 16) /*!< DTMF Support: four settings, uses two bits */
#define SIP_DTMF_RFC2833 (0 << 16) /*!< DTMF Support: RTP DTMF - "rfc2833" */
@@ -881,6 +897,7 @@
static struct sip_pvt {
ast_mutex_t lock; /*!< Dialog private lock */
int method; /*!< SIP method that opened this dialog */
+ enum invitestates invitestate; /*!< The state of the INVITE transaction only */
AST_DECLARE_STRING_FIELDS(
AST_STRING_FIELD(callid); /*!< Global CallID */
AST_STRING_FIELD(randdata); /*!< Random data */
@@ -952,8 +969,6 @@
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) */
@@ -1722,6 +1737,13 @@
ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
}
+static void sip_alreadygone(struct sip_pvt *dialog)
+{
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Setting SIP_ALREADYGONE on dialog %s\n", dialog->callid);
+ ast_set_flag(&dialog->flags[0], SIP_ALREADYGONE);
+}
+
/*! \brief returns true if 'name' (with optional trailing whitespace)
* matches the sip method 'id'.
@@ -2000,7 +2022,7 @@
ast_mutex_lock(&pkt->owner->lock);
}
if (pkt->owner->owner) {
- ast_set_flag(&pkt->owner->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(pkt->owner);
ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet.\n", pkt->owner->callid);
ast_queue_hangup(pkt->owner->owner);
ast_channel_unlock(pkt->owner->owner);
@@ -2720,17 +2742,21 @@
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);
@@ -2769,8 +2795,6 @@
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;
@@ -2929,6 +2953,7 @@
if (option_debug > 1)
ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
transmit_invite(p, SIP_INVITE, 1, 2);
+ p->invitestate = INV_CALLING;
/* Initialize auto-congest time */
p->initid = ast_sched_add(sched, p->maxtime ? (p->maxtime * 4) : SIP_TRANS_TIMEOUT, auto_congest, p);
@@ -3076,7 +3101,7 @@
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;
@@ -3091,16 +3116,17 @@
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;
@@ -3399,7 +3425,7 @@
return 0;
}
/* If the call is not UP, we need to send CANCEL instead of BYE */
- if (ast->_state == AST_STATE_RING || ast->_state == AST_STATE_RINGING) {
+ if (ast->_state == AST_STATE_RING || ast->_state == AST_STATE_RINGING || p->invitestate < INV_COMPLETED) {
needcancel = TRUE;
if (option_debug > 3)
ast_log(LOG_DEBUG, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast->_state));
@@ -3430,7 +3456,7 @@
*/
if (ast_test_flag(&p->flags[0], SIP_ALREADYGONE))
needdestroy = 1; /* Set destroy flag at end of this function */
- else
+ else if (p->invitestate != INV_CALLING)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
/* Start the process if it's not already started */
@@ -3441,7 +3467,8 @@
__sip_pretend_ack(p);
/* if we can't send right now, mark it pending */
- if (!ast_test_flag(&p->flags[0], SIP_CAN_BYE)) {
+ if (p->invitestate == INV_CALLING) {
+ /* We can't send anything in CALLING state */
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
/* Do we need a timer here if we don't hear from them at all? */
} else {
@@ -3491,6 +3518,7 @@
but we can't send one while we have "INVITE" outstanding. */
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);
+ sip_cancel_destroy(p);
}
}
}
@@ -3608,15 +3636,12 @@
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;
@@ -3748,6 +3773,7 @@
switch(condition) {
case AST_CONTROL_RINGING:
if (ast->_state == AST_STATE_RING) {
+ p->invitestate = INV_EARLY_MEDIA;
if (!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) ||
(ast_test_flag(&p->flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER)) {
/* Send 180 ringing if out-of-band seems reasonable */
@@ -3764,7 +3790,8 @@
case AST_CONTROL_BUSY:
if (ast->_state != AST_STATE_UP) {
transmit_response(p, "486 Busy Here", &p->initreq);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ p->invitestate = INV_TERMINATED;
+ sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
break;
}
@@ -3773,7 +3800,8 @@
case AST_CONTROL_CONGESTION:
if (ast->_state != AST_STATE_UP) {
transmit_response(p, "503 Service Unavailable", &p->initreq);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ p->invitestate = INV_TERMINATED;
+ sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
break;
}
@@ -3784,6 +3812,7 @@
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
transmit_response(p, "100 Trying", &p->initreq);
+ p->invitestate = INV_PROCEEDING;
break;
}
res = -1;
@@ -3792,6 +3821,7 @@
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+ p->invitestate = INV_EARLY_MEDIA;
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
break;
@@ -4311,16 +4341,19 @@
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;
}
@@ -7415,7 +7448,8 @@
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)))
@@ -7557,6 +7591,9 @@
static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
{
struct sip_request resp;
+
+ if (sipmethod == SIP_ACK)
+ p->invitestate = INV_CONFIRMED;
reqprep(&resp, p, sipmethod, seqno, newbranch);
add_header_contentLength(&resp, 0);
@@ -10341,6 +10378,7 @@
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)"));
@@ -10370,6 +10408,7 @@
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);
@@ -11644,7 +11683,7 @@
{
if (ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
/* if we can't BYE, then this is really a pending CANCEL */
- if (!ast_test_flag(&p->flags[0], SIP_CAN_BYE))
+ if (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA)
transmit_request(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, FALSE);
/* Actually don't destroy us yet, wait for the 487 on our original
INVITE, but do set an autodestruct just in case we never get it. */
@@ -11695,6 +11734,15 @@
if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 183)
resp = 183;
+ /* Any response between 100 and 199 is PROCEEDING */
+ if (resp >= 100 && resp < 200 && p->invitestate == INV_CALLING)
+ p->invitestate = INV_PROCEEDING;
+
+ /* Final response, not 200 ? */
+ if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA ))
+ p->invitestate = INV_COMPLETED;
+
+
switch (resp) {
case 100: /* Trying */
case 101: /* Dialog establishment */
@@ -11713,13 +11761,13 @@
}
}
if (find_sdp(req)) {
+ p->invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req);
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
/* Queue a progress frame only if we have SDP in 180 */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
}
- ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
@@ -11728,13 +11776,13 @@
sip_cancel_destroy(p);
/* Ignore 183 Session progress without SDP */
if (find_sdp(req)) {
+ p->invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req);
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
/* Queue a progress frame */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
}
- ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
@@ -11795,6 +11843,9 @@
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 */
@@ -11843,8 +11894,8 @@
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
}
/* If I understand this right, the branch is different for a non-200 ACK only */
+ p->invitestate = INV_TERMINATED;
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
- ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
case 407: /* Proxy authentication */
@@ -11862,7 +11913,7 @@
if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, authenticate, authorization, SIP_INVITE, 1)) {
ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", get_header(&p->initreq, "From"));
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
if (p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
}
@@ -11876,20 +11927,23 @@
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
break;
case 404: /* Not found */
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
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 */
@@ -11949,7 +12003,16 @@
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 */
@@ -12284,20 +12347,9 @@
break;
case 481: /* Call leg does not exist */
if (sipmethod == SIP_INVITE) {
- /* First we ACK */
- transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- ast_log(LOG_WARNING, "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 */
@@ -12339,7 +12391,6 @@
/* Fatal response */
if ((option_verbose > 2) && (resp != 487))
ast_verbose(VERBOSE_PREFIX_3 "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(p->sa.sin_addr));
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
@@ -12398,7 +12449,7 @@
/* ACK on invite */
if (sipmethod == SIP_INVITE)
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
if (!p->owner)
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
} else if ((resp >= 100) && (resp < 200)) {
@@ -13454,6 +13505,7 @@
if (option_debug > 1)
ast_log(LOG_DEBUG, "%s: New call is still down.... Trying... \n", c->name);
transmit_response(p, "100 Trying", req);
+ p->invitestate = INV_PROCEEDING;
ast_setstate(c, AST_STATE_RING);
if (strcmp(p->exten, ast_pickup_ext())) { /* Call to extension -start pbx on this call */
enum ast_pbx_result res;
@@ -13463,6 +13515,7 @@
switch(res) {
case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
+ p->invitestate = INV_COMPLETED;
if (ast_test_flag(req, SIP_PKT_IGNORE))
transmit_response(p, "503 Unavailable", req);
else
@@ -13470,6 +13523,7 @@
break;
case AST_PBX_CALL_LIMIT:
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
+ p->invitestate = INV_COMPLETED;
if (ast_test_flag(req, SIP_PKT_IGNORE))
transmit_response(p, "480 Temporarily Unavailable", req);
else
@@ -13497,7 +13551,7 @@
transmit_response(p, "503 Unavailable", req); /* OEJ - Right answer? */
else
transmit_response_reliable(p, "503 Unavailable", req);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
/* Unlock locks so ast_hangup can do its magic */
ast_mutex_unlock(&p->lock);
c->hangupcause = AST_CAUSE_CALL_REJECTED;
@@ -13506,6 +13560,7 @@
ast_setstate(c, AST_STATE_DOWN);
c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
}
+ p->invitestate = INV_COMPLETED;
ast_hangup(c);
ast_mutex_lock(&p->lock);
c = NULL;
@@ -13513,9 +13568,11 @@
break;
case AST_STATE_RING:
transmit_response(p, "100 Trying", req);
+ p->invitestate = INV_PROCEEDING;
break;
case AST_STATE_RINGING:
transmit_response(p, "180 Ringing", req);
+ p->invitestate = INV_PROCEEDING;
break;
case AST_STATE_UP:
if (option_debug > 1)
@@ -13601,6 +13658,7 @@
transmit_response_with_sdp(p, "200 OK", req, XMIT_CRITICAL);
}
+ p->invitestate = INV_TERMINATED;
break;
default:
ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %d\n", c->_state);
@@ -13621,6 +13679,7 @@
transmit_response(p, msg, req);
else
transmit_response_reliable(p, msg, req);
+ p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
}
@@ -13816,7 +13875,7 @@
transmit_response(p, "603 Declined (No dialog)", req);
if (!ast_test_flag(req, SIP_PKT_IGNORE)) {
append_history(p, "Xfer", "Refer failed. Outside of dialog.");
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
}
return 0;
@@ -14075,7 +14134,8 @@
{
check_via(p, req);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
+ p->invitestate = INV_CANCELLED;
if (p->owner && p->owner->_state == AST_STATE_UP) {
/* This call is up, cancel is ignored, we need a bye */
@@ -14108,12 +14168,14 @@
struct ast_channel *bridged_to;
/* If we have an INCOMING invite that we haven't answered, terminate that transaction */
- if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !ast_test_flag(req, SIP_PKT_IGNORE) && !p->owner)
+ if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !ast_test_flag(req, SIP_PKT_IGNORE) && !p->owner)
transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
+
+ p->invitestate = INV_TERMINATED;
copy_request(&p->initreq, req);
check_via(p, req);
- ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ sip_alreadygone(p);
/* Get RTCP quality before end of call */
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY) || p->owner) {
@@ -14531,7 +14593,7 @@
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;
}
@@ -14679,6 +14741,7 @@
case SIP_ACK:
/* Make sure we don't ignore this */
if (seqno == p->pendinginvite) {
+ p->invitestate = INV_CONFIRMED;
p->pendinginvite = 0;
__sip_ack(p, seqno, FLAG_RESPONSE, 0);
if (find_sdp(req)) {
@@ -14921,23 +14984,23 @@
(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);
@@ -14958,8 +15021,12 @@
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);
+ }
}
}
}
@@ -16099,6 +16166,7 @@
global_regcontext[0] = '\0';
expiry = DEFAULT_EXPIRY;
global_notifyringing = DEFAULT_NOTIFYRINGING;
+ global_limitonpeers = FALSE;
global_notifyhold = FALSE;
global_alwaysauthreject = 0;
global_allowsubscribe = FALSE;
@@ -16221,6 +16289,8 @@
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/midcomstuff/configs/extensions.conf.sample
URL: http://svn.digium.com/view/asterisk/team/oej/midcomstuff/configs/extensions.conf.sample?view=diff&rev=48310&r1=48309&r2=48310
==============================================================================
--- team/oej/midcomstuff/configs/extensions.conf.sample (original)
[... 960 lines stripped ...]
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