[svn-commits] oej: branch oej/videocaps r48257 - in
/team/oej/videocaps: ./ agi/ apps/ chan...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Tue Dec 5 03:08:32 MST 2006
Author: oej
Date: Tue Dec 5 04:08:30 2006
New Revision: 48257
URL: http://svn.digium.com/view/asterisk?view=rev&rev=48257
Log:
Update to trunk
Modified:
team/oej/videocaps/ (props changed)
team/oej/videocaps/.cleancount
team/oej/videocaps/agi/Makefile
team/oej/videocaps/apps/app_voicemail.c
team/oej/videocaps/channels/chan_sip.c
team/oej/videocaps/configs/voicemail.conf.sample
team/oej/videocaps/doc/snmp.txt
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 Tue Dec 5 04:08:30 2006
@@ -1,1 +1,1 @@
-/trunk:1-48207
+/trunk:1-48256
Modified: team/oej/videocaps/.cleancount
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/.cleancount?view=diff&rev=48257&r1=48256&r2=48257
==============================================================================
--- team/oej/videocaps/.cleancount (original)
+++ team/oej/videocaps/.cleancount Tue Dec 5 04:08:30 2006
@@ -1,1 +1,1 @@
-26
+27
Modified: team/oej/videocaps/agi/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/agi/Makefile?view=diff&rev=48257&r1=48256&r2=48257
==============================================================================
--- team/oej/videocaps/agi/Makefile (original)
+++ team/oej/videocaps/agi/Makefile Tue Dec 5 04:08:30 2006
@@ -21,7 +21,7 @@
include $(ASTTOPDIR)/Makefile.rules
-all: #$(AGIS)
+all: $(AGIS)
strcompat.c: ../main/strcompat.c
@cp $< $@
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=48257&r1=48256&r2=48257
==============================================================================
--- team/oej/videocaps/apps/app_voicemail.c (original)
+++ team/oej/videocaps/apps/app_voicemail.c Tue Dec 5 04:08:30 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"
@@ -1099,6 +1100,7 @@
goto yuck;
}
if (!strcasecmp(coltitle, "recording")) {
+ off_t offset;
res = SQLGetData(stmt, x + 1, SQL_BINARY, NULL, 0, &colsize2);
fdlen = colsize2;
if (fd > -1) {
@@ -1109,24 +1111,27 @@
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);
@@ -1147,8 +1152,6 @@
yuck:
if (f)
fclose(f);
- if (fdm)
- munmap(fdm, fdlen);
if (fd > -1)
close(fd);
return x - 1;
@@ -8003,6 +8006,8 @@
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Saving message as is\n");
ast_stream_and_wait(chan, "vm-msgsaved", "");
+ STORE(recordfile, vmu->mailbox, vmu->context, -1, chan, vmu, fmt, duration, vms);
+ DISPOSE(recordfile, -1);
cmd = 't';
return res;
}
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=48257&r1=48256&r2=48257
==============================================================================
--- team/oej/videocaps/channels/chan_sip.c (original)
+++ team/oej/videocaps/channels/chan_sip.c Tue Dec 5 04:08:30 2006
@@ -241,15 +241,15 @@
\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, /*!< Invite sent, no answer */
- INV_PROCEEDING, /*!< We got 1xx message */
- INV_EARLY_MEDIA, /*!< We got 18x message with to-tag back */
- INV_COMPLETED, /*!< Got final response with error. Wait for ACK, then CONFIRMED */
- INV_CONFIRMED, /*!< Confirmed response - we've got an ack (Incoming calls only) */
- INV_TERMINATED, /*!< 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 /*!< Transaction cancelled by client or server in non-terminated state */
+ 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;
@@ -1690,6 +1690,14 @@
ast_verbose("Initreq: %d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
}
+/*! \brief Encapsulate setting of SIP_ALREADYGONE to be able to trace it with debugging */
+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'.
@@ -1968,7 +1976,7 @@
sip_pvt_lock(pkt->owner);
}
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);
@@ -3463,7 +3471,7 @@
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 (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));
@@ -3484,7 +3492,7 @@
*/
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 */
@@ -3553,6 +3561,7 @@
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);
}
}
}
@@ -3872,6 +3881,7 @@
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 */
@@ -3888,7 +3898,8 @@
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;
}
@@ -3897,7 +3908,8 @@
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;
}
@@ -3908,6 +3920,7 @@
!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;
@@ -3916,6 +3929,7 @@
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;
@@ -12705,7 +12719,7 @@
if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, 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);
}
@@ -12719,20 +12733,23 @@
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 */
@@ -12785,7 +12802,16 @@
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 */
@@ -13145,21 +13171,9 @@
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 */
@@ -13201,7 +13215,6 @@
/* 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 */
@@ -13260,7 +13273,7 @@
/* 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)) {
@@ -14349,7 +14362,7 @@
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 */
sip_pvt_unlock(p);
c->hangupcause = AST_CAUSE_CALL_REJECTED;
@@ -14673,7 +14686,7 @@
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;
@@ -14932,7 +14945,7 @@
{
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) {
@@ -14975,7 +14988,7 @@
if (sipdebug && option_debug)
ast_log(LOG_DEBUG, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
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) {
Modified: team/oej/videocaps/configs/voicemail.conf.sample
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/configs/voicemail.conf.sample?view=diff&rev=48257&r1=48256&r2=48257
==============================================================================
--- team/oej/videocaps/configs/voicemail.conf.sample (original)
+++ team/oej/videocaps/configs/voicemail.conf.sample Tue Dec 5 04:08:30 2006
@@ -87,6 +87,12 @@
;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/videocaps/doc/snmp.txt
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/doc/snmp.txt?view=diff&rev=48257&r1=48256&r2=48257
==============================================================================
--- team/oej/videocaps/doc/snmp.txt (original)
+++ team/oej/videocaps/doc/snmp.txt Tue Dec 5 04:08:30 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/videocaps/sounds/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/sounds/Makefile?view=diff&rev=48257&r1=48256&r2=48257
==============================================================================
--- team/oej/videocaps/sounds/Makefile (original)
+++ team/oej/videocaps/sounds/Makefile Tue Dec 5 04:08:30 2006
@@ -52,7 +52,10 @@
MM:=$(subst -G722,-g722,$(MM))
MOH:=$(MM:MOH-%=asterisk-moh-%.tar.gz)
MOH_TAGS:=$(MM:MOH-%=$(MOH_DIR)/.asterisk-moh-%)
+# If "fetch" is used, --continue is not a valid option.
+ifeq ($(WGET),wget)
WGET_ARGS:=--continue
+endif
all: $(CORE_SOUNDS) $(EXTRA_SOUNDS) $(MOH)
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