[asterisk-commits] murf: branch murf/bug8221-1.4 r48288 - in
/team/murf/bug8221-1.4: ./ apps/ ch...
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Dec 5 18:44:06 MST 2006
Author: murf
Date: Tue Dec 5 19:44:05 2006
New Revision: 48288
URL: http://svn.digium.com/view/asterisk?view=rev&rev=48288
Log:
Merged revisions 48088,48095,48101,48105,48107,48113,48115,48129,48135,48143,48147,48152,48155,48158,48162,48166,48168,48177,48179,48184-48186,48190,48193,48195,48199,48219,48223,48228,48230,48234,48237,48247-48248,48252,48254,48264,48268,48270,48279,48281 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r48088 | file | 2006-11-28 09:57:16 -0700 (Tue, 28 Nov 2006) | 10 lines
Merged revisions 48087 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov 2006) | 2 lines
According to the research I have done we never needed to include compiler.h in the first place so let's not! (issue #8430 reported by edguy3)
........
................
r48095 | qwell | 2006-11-28 11:26:53 -0700 (Tue, 28 Nov 2006) | 2 lines
Export several more variables in top level Makefile. Inspired by issue 8438.
................
r48101 | file | 2006-11-28 21:26:53 -0700 (Tue, 28 Nov 2006) | 2 lines
Don't crash if the mailstream was not created.
................
r48105 | oej | 2006-11-29 01:03:36 -0700 (Wed, 29 Nov 2006) | 2 lines
Clarify RTP timers. Sorry, grandma.
................
r48107 | file | 2006-11-29 09:50:33 -0700 (Wed, 29 Nov 2006) | 10 lines
Merged revisions 48106 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2 lines
If the frame was duplicated before writing out then we need to free it. (issue #8429 reported by edguy3)
........
................
r48113 | oej | 2006-11-29 12:56:56 -0700 (Wed, 29 Nov 2006) | 14 lines
Explain the use device status system implemented in SIP for subscriptions,
queues and manager a bit better.
Like in 1.2, you will get more detailed information if you set a call
limit for a device. When the call limit is reached, the status system will
report a device as busy.
For queues, setting a call limit per SIP device is propably a requirement.
In most cases, it will work much better if you only use type=peer and not
type=friend. We might decide to backport the new setting from trunk to
apply all call limits to the peer part of a friend only.
................
r48115 | file | 2006-11-29 14:05:17 -0700 (Wed, 29 Nov 2006) | 2 lines
Use MAILTMPLEN instead of sizeof in mm_login. (issue #8420 reported by slimey)
................
r48129 | oej | 2006-11-30 02:12:30 -0700 (Thu, 30 Nov 2006) | 3 lines
Clean up bad dialogs properly. Caused by GS 487 adapter without CSEQ on separate line
in the REGISTER request. Imported from 1.2.
................
r48135 | oej | 2006-11-30 07:29:39 -0700 (Thu, 30 Nov 2006) | 3 lines
Explain status reports and
make codefreeze more happy :-)
................
r48143 | file | 2006-11-30 10:57:35 -0700 (Thu, 30 Nov 2006) | 10 lines
Merged revisions 48142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2 lines
Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage)
........
................
r48147 | file | 2006-11-30 11:19:55 -0700 (Thu, 30 Nov 2006) | 9 lines
Blocked revisions 48146 via svnmerge
........
r48146 | file | 2006-11-30 13:17:54 -0500 (Thu, 30 Nov 2006) | 2 lines
Remember the pointer to the allocated block of memory so that we can free it and not cause a memory leak. (issue #8449 reported by arkadia)
........
................
r48152 | file | 2006-11-30 11:47:40 -0700 (Thu, 30 Nov 2006) | 10 lines
Merged revisions 48151 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2 lines
Print certain CDR messages out at the NOTICE level versus WARNING since they can occur when used with the CDR applications and are perfectly fine. (issue #8367 reported by dartvader)
........
................
r48155 | file | 2006-11-30 12:05:14 -0700 (Thu, 30 Nov 2006) | 10 lines
Merged revisions 48154 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2 lines
Do not listen for DTMF on the bridge that comes into existence when ParkedCall is executed. This means native bridging can now occur for this. (issue #8406 reported by kebl0155)
........
................
r48158 | file | 2006-11-30 13:07:55 -0700 (Thu, 30 Nov 2006) | 10 lines
Merged revisions 48157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2 lines
Only print out debug message if bridged channel is not NULL. (issue #8412 reported by jubilex)
........
................
r48162 | file | 2006-11-30 13:28:19 -0700 (Thu, 30 Nov 2006) | 9 lines
Blocked revisions 48161 via svnmerge
........
r48161 | file | 2006-11-30 15:27:29 -0500 (Thu, 30 Nov 2006) | 2 lines
Don't write AST_FRAME_NULL or AST_FRAME_IAX frames out to the channel driver. (issue #8390 reported by hselasky)
........
................
r48166 | oej | 2006-11-30 13:51:04 -0700 (Thu, 30 Nov 2006) | 2 lines
Issue 8319 - change noncecount before using it.
................
r48168 | file | 2006-11-30 14:18:24 -0700 (Thu, 30 Nov 2006) | 2 lines
Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan)
................
r48177 | oej | 2006-12-01 10:41:56 -0700 (Fri, 01 Dec 2006) | 5 lines
- Backport of the "limitonpeers" patch from trunk, to fix a lot of issues with queues and SIP device states
- Remove support for T.38 early media, since it's impossible.
(Two patches in one - extra friday evening offer due to being off line from svn today... :-)
................
r48179 | tilghman | 2006-12-01 12:38:59 -0700 (Fri, 01 Dec 2006) | 2 lines
Double-unlock error (reported by blitzrage on IRC)
................
r48184 | qwell | 2006-12-01 13:19:44 -0700 (Fri, 01 Dec 2006) | 6 lines
Merged revisions 48182 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
........
................
r48185 | qwell | 2006-12-01 13:24:32 -0700 (Fri, 01 Dec 2006) | 2 lines
erm, hmm...I typod a merge...that went...strangely
................
r48186 | qwell | 2006-12-01 13:25:51 -0700 (Fri, 01 Dec 2006) | 10 lines
Merged revisions 48183 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 lines
Fix a small typo - issue 8848, reported by pabelanger
........
................
r48190 | russell | 2006-12-01 16:16:28 -0700 (Fri, 01 Dec 2006) | 12 lines
FreeBSD 6.1 does not include wget by default. However, it has fetch which will
work just fine for our purposes of downloading the sounds packages. So, check
for both wget and fetch and the configure script and use what was found to
download them. If neither one was found, and sound packages are selected that
must be downloaded, the install process will print out an informative error
message indicating the situation.
Also, fix a couple places where "make" was hard coded into some output messages
by replacing them with the $(MAKE) variable.
(issue #8451, initial patch by pabelanger, with additional modifications by me)
................
r48193 | kpfleming | 2006-12-01 16:37:28 -0700 (Fri, 01 Dec 2006) | 10 lines
Merged revisions 48192 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006) | 2 lines
if Dial() is going to send music-on-hold to the calling party, it has to send PROGRESS first to ensure that the reverse audio path has been setup first (BE-106)
........
................
r48195 | russell | 2006-12-01 20:50:58 -0700 (Fri, 01 Dec 2006) | 3 lines
Backport the comment containing the warning regarding the limitations on the
usage of this function. It is thread safe, but not technically reentrant.
................
r48199 | oej | 2006-12-02 04:32:51 -0700 (Sat, 02 Dec 2006) | 11 lines
- Disable RTP hold timers while T.38 fax transmission happens
- Encapsulate RTP timers in the rtp structure so we have one for video and one for audio
The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send
something that video phones support in the RTP stream.
I now this is a big architectual change at this stage for 1.4, but decided it was needed
to avoid future bug reports.
- Document the RTP NAT keepalive option in sip.conf.sample
Issue 7679 in the bug tracker. Please test.
................
r48219 | oej | 2006-12-02 14:45:58 -0700 (Sat, 02 Dec 2006) | 3 lines
- Removing one of two pieces of code to handle 481 response on INVITE
- Move handling of REFER response to handle_response_refer()
................
r48223 | russell | 2006-12-02 23:34:14 -0700 (Sat, 02 Dec 2006) | 3 lines
When "fetch" is in use, instead of "wget", --continue is not a valid option.
(issue #8451)
................
r48228 | qwell | 2006-12-04 10:43:24 -0700 (Mon, 04 Dec 2006) | 4 lines
Attempt to document some of the dependencies that are needed for net-snmp
Issue 8499 - initial patch by blitzrage.
................
r48230 | qwell | 2006-12-04 10:54:46 -0700 (Mon, 04 Dec 2006) | 4 lines
Add documentation to voicemail.conf.sample for ODBC storage.
Issue 8499 - patch by blitzrage.
................
r48234 | file | 2006-12-04 11:16:31 -0700 (Mon, 04 Dec 2006) | 9 lines
Blocked revisions 48233 via svnmerge
........
r48233 | file | 2006-12-04 13:14:46 -0500 (Mon, 04 Dec 2006) | 2 lines
If the generic bridge tells us not to retry, and we have a frame to spit out then break the bridge. Props to markit in #asterisk-bugs for bringing this up.
........
................
r48237 | qwell | 2006-12-04 12:08:18 -0700 (Mon, 04 Dec 2006) | 11 lines
Blocked revisions 48236 via svnmerge
........
r48236 | qwell | 2006-12-04 13:06:26 -0600 (Mon, 04 Dec 2006) | 4 lines
Fix an issue where a message isn't saved correctly when using ODBC storage and reviewing a message.
Issue 8016 - patch by sokhapkin.
........
................
r48247 | qwell | 2006-12-04 14:23:11 -0700 (Mon, 04 Dec 2006) | 14 lines
Blocked revisions 48246 via svnmerge
........
r48246 | qwell | 2006-12-04 15:20:34 -0600 (Mon, 04 Dec 2006) | 7 lines
Revert change from 8016 - this breaks other stuff... Needs further review.
Tip: When you've reported a bug about something and somebody has put up a
patch for it.. It's not a good idea to open a completely new bug and say that
something is broken because of the patch in the other bug - PLEASE mention
something in the bug where the patch was actually created.
........
................
r48248 | qwell | 2006-12-04 14:48:41 -0700 (Mon, 04 Dec 2006) | 2 lines
Fix an issue which didn't allow unavail/greet/busy/etc messages from being saved into ODBC (and probably IMAP).
................
r48252 | tilghman | 2006-12-04 18:34:34 -0700 (Mon, 04 Dec 2006) | 14 lines
Merged revisions 48251 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006) | 6 lines
If the recording in the database is too large, it will fail to retrieve with
an mmap error. Not too sure why this doesn't happen when we put it in the
database, also, but since that doesn't seem to be broken, I'm not going to fix
it (at least until someone reports it). Solution is to ask for the file in
smaller chunks. (Bug 8385)
........
................
r48254 | tilghman | 2006-12-04 18:41:02 -0700 (Mon, 04 Dec 2006) | 2 lines
Oops, forgot to release the odbc handle
................
r48264 | oej | 2006-12-05 05:39:30 -0700 (Tue, 05 Dec 2006) | 11 lines
Updating sip.conf.sample with information about T38 not working
when chan_local or chan_agent is involved in the call.
I don't know how big a fix that would be to solve, but this is
the current state of affairs.
(Chan_sip currently checks if the other side of the bridge
has a SIP tech. We could/should implement another check,
possibly for udptl_write or some flag in the ast_channel
structure).
................
r48268 | oej | 2006-12-05 08:59:05 -0700 (Tue, 05 Dec 2006) | 2 lines
Add missing s from another repository. (thanks jcmoore!)
................
r48270 | oej | 2006-12-05 10:29:43 -0700 (Tue, 05 Dec 2006) | 4 lines
Merging the invitestate-1.4 branch after successful testing.
Will check if I can solve this with less changes in 1.2.
................
r48279 | qwell | 2006-12-05 13:42:52 -0700 (Tue, 05 Dec 2006) | 4 lines
Fix curl version number testing to be much more friendly to non-bash shells.
Issue 8508, patch by me. This *SHOULD* be POSIX compliant now..
................
r48281 | file | 2006-12-05 13:45:28 -0700 (Tue, 05 Dec 2006) | 2 lines
Regenerate configure from Qwell's last commit.
................
Modified:
team/murf/bug8221-1.4/ (props changed)
team/murf/bug8221-1.4/Makefile
team/murf/bug8221-1.4/apps/app_dial.c
team/murf/bug8221-1.4/apps/app_voicemail.c
team/murf/bug8221-1.4/channels/chan_gtalk.c
team/murf/bug8221-1.4/channels/chan_iax2.c
team/murf/bug8221-1.4/channels/chan_phone.c
team/murf/bug8221-1.4/channels/chan_sip.c
team/murf/bug8221-1.4/configs/extensions.conf.sample
team/murf/bug8221-1.4/configs/sip.conf.sample
team/murf/bug8221-1.4/configs/voicemail.conf.sample
team/murf/bug8221-1.4/configure
team/murf/bug8221-1.4/configure.ac
team/murf/bug8221-1.4/doc/manager.txt
team/murf/bug8221-1.4/doc/snmp.txt
team/murf/bug8221-1.4/include/asterisk/rtp.h
team/murf/bug8221-1.4/include/asterisk/utils.h
team/murf/bug8221-1.4/main/cdr.c
team/murf/bug8221-1.4/main/cli.c
team/murf/bug8221-1.4/main/rtp.c
team/murf/bug8221-1.4/makeopts.in
team/murf/bug8221-1.4/res/res_features.c
team/murf/bug8221-1.4/sounds/Makefile
Propchange: team/murf/bug8221-1.4/
------------------------------------------------------------------------------
Binary property 'branch-1.2-blocked' - no diff available.
Propchange: team/murf/bug8221-1.4/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.
Propchange: team/murf/bug8221-1.4/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Dec 5 19:44:05 2006
@@ -1,1 +1,1 @@
-/branches/1.4:1-48065
+/branches/1.4:1-48287
Modified: team/murf/bug8221-1.4/Makefile
URL: http://svn.digium.com/view/asterisk/team/murf/bug8221-1.4/Makefile?view=diff&rev=48288&r1=48287&r2=48288
==============================================================================
--- team/murf/bug8221-1.4/Makefile (original)
+++ team/murf/bug8221-1.4/Makefile Tue Dec 5 19:44:05 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/murf/bug8221-1.4/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/murf/bug8221-1.4/apps/app_dial.c?view=diff&rev=48288&r1=48287&r2=48288
==============================================================================
--- team/murf/bug8221-1.4/apps/app_dial.c (original)
+++ team/murf/bug8221-1.4/apps/app_dial.c Tue Dec 5 19:44:05 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/murf/bug8221-1.4/apps/app_voicemail.c
URL: http://svn.digium.com/view/asterisk/team/murf/bug8221-1.4/apps/app_voicemail.c?view=diff&rev=48288&r1=48287&r2=48288
==============================================================================
--- team/murf/bug8221-1.4/apps/app_voicemail.c (original)
+++ team/murf/bug8221-1.4/apps/app_voicemail.c Tue Dec 5 19:44:05 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/murf/bug8221-1.4/channels/chan_gtalk.c
URL: http://svn.digium.com/view/asterisk/team/murf/bug8221-1.4/channels/chan_gtalk.c?view=diff&rev=48288&r1=48287&r2=48288
==============================================================================
--- team/murf/bug8221-1.4/channels/chan_gtalk.c (original)
+++ team/murf/bug8221-1.4/channels/chan_gtalk.c Tue Dec 5 19:44:05 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/murf/bug8221-1.4/channels/chan_iax2.c
URL: http://svn.digium.com/view/asterisk/team/murf/bug8221-1.4/channels/chan_iax2.c?view=diff&rev=48288&r1=48287&r2=48288
==============================================================================
--- team/murf/bug8221-1.4/channels/chan_iax2.c (original)
+++ team/murf/bug8221-1.4/channels/chan_iax2.c Tue Dec 5 19:44:05 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/murf/bug8221-1.4/channels/chan_phone.c
URL: http://svn.digium.com/view/asterisk/team/murf/bug8221-1.4/channels/chan_phone.c?view=diff&rev=48288&r1=48287&r2=48288
==============================================================================
--- team/murf/bug8221-1.4/channels/chan_phone.c (original)
+++ team/murf/bug8221-1.4/channels/chan_phone.c Tue Dec 5 19:44:05 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/murf/bug8221-1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/murf/bug8221-1.4/channels/chan_sip.c?view=diff&rev=48288&r1=48287&r2=48288
==============================================================================
--- team/murf/bug8221-1.4/channels/chan_sip.c (original)
+++ team/murf/bug8221-1.4/channels/chan_sip.c Tue Dec 5 19:44:05 2006
@@ -244,6 +244,21 @@
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
@@ -513,6 +528,7 @@
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 */
@@ -702,7 +718,7 @@
#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" */
@@ -876,6 +892,7 @@
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 */
@@ -947,8 +964,6 @@
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) */
@@ -1594,6 +1609,13 @@
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'.
@@ -1872,7 +1894,7 @@
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);
@@ -2592,17 +2614,21 @@
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);
@@ -2641,8 +2667,6 @@
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;
@@ -2801,6 +2825,7 @@
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);
@@ -2944,7 +2969,7 @@
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;
@@ -2959,16 +2984,17 @@
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;
@@ -3267,7 +3293,7 @@
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));
@@ -3291,7 +3317,7 @@
*/
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 */
@@ -3302,7 +3328,8 @@
__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 {
@@ -3352,6 +3379,7 @@
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);
}
}
}
@@ -3469,15 +3497,12 @@
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;
@@ -3609,6 +3634,7 @@
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 */
@@ -3625,7 +3651,8 @@
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;
}
@@ -3634,7 +3661,8 @@
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;
}
@@ -3645,6 +3673,7 @@
!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;
@@ -3653,6 +3682,7 @@
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;
@@ -4172,16 +4202,19 @@
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;
}
@@ -7228,7 +7261,8 @@
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)))
@@ -7370,6 +7404,9 @@
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);
@@ -10154,6 +10191,7 @@
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)"));
@@ -10183,6 +10221,7 @@
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);
@@ -11457,7 +11496,7 @@
{
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. */
@@ -11508,6 +11547,15 @@
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 */
@@ -11526,13 +11574,13 @@
}
}
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;
@@ -11541,13 +11589,13 @@
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;
@@ -11597,6 +11645,9 @@
if (bridgepvt->udptl) {
if (p->t38.state == T38_PEER_REINVITE) {
sip_handle_t38_reinvite(bridgepeer, p, 0);
[... 1127 lines stripped ...]
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