Kinsey Moore [Mon, 13 Jun 2011 14:30:51 +0000 (14:30 +0000)]
ConfBridge: Use of bridge or user profiles that don't exist
Bridge and user profiles are not checked for existence before use. The lack
of a fully formed bridge profile can cause a segfault when sounds are accessed.
This change ensures that bridge and user profiles exist prior to usage
attempts.
Explicity check the last entry in the DB and make sure that we don't iterate
past it. Since there can be no duplicates, this just makes sure that we stop
after matching the last key.
This patch also refactors the code to get away from some code duplication. A
previous patch added many astdb tests and this patch passed them.
Deadlock is possible in ast_do_pickup() when holding the target channel
lock and trying to get the chan channel lock. Also, holding the target
lock when calling ast_channel_masquerade() is not a good idea because that
routine does deadlock avoidance.
* Removed the need to hold the target lock after marking the target with a
datastore and getting the connected line data off of the target channel.
* Moved can_pickup() to ast_can_pickup() in features.c. Now all the call
pickup methods use the same basic call pickup availability check.
Adds ast_escape_encoded utility to properly handle escaping of quoted field before uri.
This commit backports a feature in trunk affecting initreqprep so that display name won't
be encoded improperly. Also includes unit tests for the ast_escape_quoted function.
This patch gives 1.8 a much improved outlook in countries which don't use standard
ASCII characters.
Damien Wedhorn [Thu, 9 Jun 2011 11:05:07 +0000 (11:05 +0000)]
Add autoanswer to skinny.
Autoanswer added to skinny based on incoming chan var SKINNY_AUTOANSWER.
Initial value must be the time to autoanswer in ms, then optionally :BEEP
to play a tone when answered and :MUTE to mute the mic when answering.
eg 3000:MUTE:BEEP will ring for 3 secs, then answer, mute the mic, and
play a beep. just 3000 would answer afer 3 secs of ringing with no
beep and full two way audio.
Ring all queue with more than 255 agents will cause crash.
1. Create a ring-all queue with 500 permanent agents.
2. Call it.
3. Asterisk will crash.
The watchers array in app_queue.c has a hard limit of 255. Bounds
checking is not done on this array. No sane person should put 255 people
in a ring-all queue, but we should not crash anyway.
Damien Wedhorn [Wed, 8 Jun 2011 11:38:56 +0000 (11:38 +0000)]
Remove skinny do_monitor and use ast_sched_start instead
The do_monitor seemed to be there for task scheduling and network monitoring. However, the network monitoring has a dedicated thread so the ast_io_wait was basically just a usleep as it didn't actually seem to be monitoring anything.
Make handle_request_publish do dialog expiration and destruction.
This patch fixes handle_request_publish so that it does dialog expiration and destruction.
Without this patch the incoming PUBLISH requests will get stuck in the dialog list.
Restarting asterisk is the only way to remove them.
Personal observation on one system the server hung up while looping through the channels
rendering asterisk unusable and all sip phones unregisterd when they try reregister
more requests are added.
Asterisk crash when unloading cdr_radius/cel_radius.
The rc_openlog() API call is passed a string that is used by openlog() to
format log messages. The openlog() does not copy the string it just keeps
a pointer to it. When the module is unloaded, the string is gone from
memory. Depending upon module load order and if the other module then has
an error, a crash happens.
* Pass rc_openlog() a strdup'd string with the understanding that there
will be a small memory leak if the cdr_radius/cel_radius modules are
unloaded.
* Call rc_destroy() to free the rc handle memory when the module is
unloaded.
Must commit the subscription fixes together with the integration
subscription tests. The subscription fixes cause an erroneously passing
test to fail. The new subscription tests detect errors without the
subscription fixes.
* Added missing event_names[] table entry.
* Reworked ast_event_check_subscriber()/match_sub_ie_val_to_event() to
correctly detect if a subscriber exists for the proposed event.
* Made match_ie_val() and match_sub_ie_val_to_event() check the buffer
length for RAW payload types.
* Fixed error handling memory leak in ast_event_sub_activate(),
ast_event_unsubscribe(), and ast_event_queue().
* Made ast_event_new() and ast_event_check_subscriber() better protect
themselves from an invalid payload type.
* Added container lock protection between removing old cache events and
adding the new cached event in
ast_event_queue_and_cache()/event_update_cache().
Russell Bryant [Wed, 1 Jun 2011 21:31:40 +0000 (21:31 +0000)]
Support routing text messages outside of a call.
Asterisk now has protocol independent support for processing text messages
outside of a call. Messages are routed through the Asterisk dialplan.
SIP MESSAGE and XMPP are currently supported. There are options in sip.conf
and jabber.conf that enable these features.
There is a new application, MessageSend(). There are two new functions,
MESSAGE() and MESSAGE_DATA(). Documentation will be available on
the project wiki, wiki.asterisk.org.
Thanks to Terry Wilson for the assistance with development and to David Vossel
for helping with some additional testing.
Alexandr Anikin [Wed, 1 Jun 2011 10:45:12 +0000 (10:45 +0000)]
Merged revisions 321528 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r321528 | may | 2011-06-01 14:40:19 +0400 (Wed, 01 Jun 2011) | 14 lines
Fix double alerting, add forced alerting before answer
Fix double alerting (it wasn't fixed here by issue #18542)
Add forced alerting before connect (if it wasn't before)
Try to send all packets from outgoing queue rather than one only
Call goes into clearing state when disconnect command is received
This patch removes all of the unnecessary deadlock
avoidance loops that occur in chan_local. It also
resolves an issue with a deadlock triggered by
local channel optimizations.
Crash when using hagi and no servers are available.
When none of the servers returned by the SRV querey respond, asterisk
crashes. The problem is that if the loop over all the SRV entries
finishes then the srv_context has already been cleaned up.
* Make ast_srv_cleanup() check to see if the context is already cleaned
up.
Fix *8 directed pickup locks system during pickupsound play out
move playout from sip_pickup_thread to bridge using BRIDGE_PLAY_SOUND method,
This stop the clash of 2 threads trying to write audio to same channel.
In addition fixes choppy audio beep in issue 19177.
Fixed build problem with dev mode enabled, which was caused by commit 321100. Reformulated patch to be more generic.
Moved the sip uri parse variable initalization to parse_uri_full in reqresp_parser.c. This will ensure that any use of parse uri will have null output variables if the parse fails.
ast_sockaddr_resolve() in netsock2.c may deref a null pointer
Added a null check in netsock2 ast_sockaddr_resolve() as well as added default initalizers in chan_sip parse_uri_legacy_check() to make sure that invalid uris will make null (and not undefined) user,pass,domain,transport variables
Initialize stack-allocated ast_sockaddrs before use
It is important to always initialize ast_sockaddrs before use--even if they
are passed to ast_sockaddr_copy as the underlying storage could be bigger
than what ends up being copied--leaving part of the data unitialized.
........
Terry Wilson [Thu, 26 May 2011 15:55:22 +0000 (15:55 +0000)]
Use va_copy for stringfields
The ast_string_field_build_va functions were written to take to separate
va_lists to work around FreeBSD 4 not having va_copy defined.
In the end, we don't support anything using gcc < 3 anyway because we use
va_copy all over the place anyway. This patch just simplifies things by
removing the second va_list function arguments in favor of va_copy.
Review: https://reviewboard.asterisk.org/r/1233/
--This line, and those below, will be ignored--
M include/asterisk/stringfields.h
M main/utils.c
M main/channel.c
Native SIP CCSS sends bad CC cancel SUBSCRIBE message.
The SUBSCRIBE message used to cancel a CC request has incorrect To/From
SIP headers. They are reversed and the dialog tags are the same when they
should not be. If pedantic mode was disabled, then the cancel would have
succeeded despite the incorrect message.
* The SIP_OUTGOING flag was not set correctly for the dialog and I had to
move some CC subscribe handling code as a result.
* Initialized the dialog subscribed type to CALL_COMPLETION earlier. If a
CC request SUBSCRIBE message comes in and the CC instance is not found,
the 404 response was duplicated.
The AMI Newstate event contains different information between v1.4 and v1.8.
The addition of connected line support in v1.8 changes the behavior of the
channel caller ID somewhat. The channel caller ID value no longer time
shares with the connected line ID on outgoing call legs. The timing of
some AMI events/responses output the connected line ID as caller ID.
These party ID's are now separate.
* The ConnectedLineNum and ConnectedLineName headers were added to many
AMI events/responses if the CallerIDNum/CallerIDName headers were also
present.
Recent crashes from zombie channels suggests that they need a safe home to
goto. When a masquerade happens, the physical part of the zombie channel
is hungup. The hangup normally sets the channel private pointer to NULL.
If someone then blindly does a callback to the channel driver, a crash is
likely because the private pointer is NULL.
The masquerade now sets the channel technology of zombie channels to the
kill channel driver.
Related to the following issues:
(issue #19116)
(issue #19310)
Add ConnectedLineNum/Name headers to output of AMI action Status.
* Add ConnectedLineNum and ConnectedLineName headers to the output of the
AMI action Status. This makes it easier to find out who the channel is
connected to without having to lookup BridgedChannel or when they are
connected to an application (e.g.: VoiceMail) which has no bridged
channel.
* Bridged channels with no CallerID had "" instead of "<unknown>" output,
that might be a bug as "<unknown>" was what older versions used.
Don't generate spurious "No: command not found" messages when running the
configure script on a system that has neither gmime-config nor pkg-config.
........
The meetme CLI command completion leaves conferences mutex locked.
When issuing a meetme kick CLI command and an invalid (non-existent)
conference number is specified, pressing Tab leaves the conferences mutex
locked and, therefore, all conferences deadlock.
This commit modifies the way polling is done on TLS sockets.
Because of the buffering the TLS layer does, polling is unreliable. If poll is
called while there is data waiting to be read in the TLS layer but not at the
network layer, the messaging processing engine will not proceed until something
else writes data to the socket, which may not occur. This change modifies the
logic around TLS sockets to only poll after a failed read on a non-blocking
socket. This way we know that there is no data waiting to be read from the
buffering layer.
(closes issue #19182)
Reported by: st
Patches:
ssl-poll-fix3.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
........
imapfolders being set in the general section of voicemail would cause the inbox folder name to
change. Since sound file names are made based on the names of the folders, this would cause
the audio related to that folder name to change and if Asterisk attempted to play it, the
channel would instantly hang up when the audio file couldn't be found. This patch searches for
the name of the folder first to leave existing behavior in tact and if that fails, it uses
the normal inbox name to get the sound file instead.
Crash while transferring a call during DTMF feature timeout.
When a call is being attended transferred during the time between
AST_FRAME_DTMF_BEGIN and AST_FRAME_DTMF_END, the transferred channel
becomes a zombie (so tech data is not available), making ast_dtmf_stream()
segfault when it tries to send the DTMF digit (at least with SIP
channels).
Patch based on feature-end-zombie.patch uploaded by Irontec (license 1256)
* Check for zombies when ast_channel_bridge() returns.
* Guarantee that the fo parameter value is initialized in
ast_channel_bridge() before any returns.
The directed pickup applications can cause a crash if the pickup was
successful because the dialplan keeps executing.
This patch does the following:
* Completes the channel masquerade on a successful pickup before the
application returns. The channel is now guaranteed a zombie and must not
continue executing the dialplan.
* Changes the return value of the directed pickup applications to return
zero if the pickup failed and nonzero(-1) if the pickup succeeded.
* Made some code optimizations that no longer require re-checking the
pickup channel to see if it is still available to pickup.
Adds legacy_useroption_parsing to address interoperability concerns.
With the new option engaged, Asterisk should interpret user fields with useroptions
contained within the userfield of the uri by stripping them out of the original message
whenever a semicolon is encountered in the userfield string.
CCSS generic agent with POTS and ISDN phones fail caller busy call-back test.
If the following is true after a CCSS activation:
* The generic agent is for an analog phone or ISDN phone. (Caller party)
* The called party becomes available.
* The caller party is not available.
When the caller party becomes available, the caller is not alerted to the
called party being available. The generic agent still thinks the caller
is busy.
* Fixed the generic agent device state event subscription to look for all
device states that are considered available.
* Encapsulated the device state test for CCSS generic device available in
cc_generic_is_device_available(). Made the generic agent and monitor use
the new function instead of the manually coded inline equivalent.
Make sure everyone gets an unhold when a transfer succeeds
Some phones, like the Snom phones, send a hold to the transfer target after
before sending the REFER. We need to make sure that we unhold the parties
that are being connected after the masquerade. If Local channels with the /nm
option are used when dialing the parties, hold music would still be playing on
the transfer target, even after being connected with the transferee.
........
................
................
The fix for issue 18882 broke retrieving non-realtime peers from the ast_db
on restart/reload. This patch tries to unbreak things while leaving the intent
of the original fix intact.
(closes issue #19318)
Reported by: remiq
Patches:
diff.txt uploaded by twilson (license 396)
Tested by: lmadsen, remiq
........
The mISDN HDLC mode is prevented on dialed channels.
The use of mISDN HDLC mode is prevented if the mISDN dial technology
option 'h1' is used when config option astdtmf=yes.
There is a bug in channels/misdn/isdn_lib.c which prevents the use of HDLC
mode. Instead of setting the channel to HDLC mode it is set to
transparent(no dsp, no hdlc), although hdlc is not "no hdlc". I.e the
logging message is correct, but the if condition is not.
Richard Mudgett [Tue, 17 May 2011 20:13:27 +0000 (20:13 +0000)]
Option needed for Q931_IE_TIME_DATE to be optional in CONNECT message.
The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG.
Add option to specify if and how much of the current time is put in
Q931_IE_TIME_DATE.
* Send date/time ie never.
* Send date/time ie date only.
* Send date/time ie date and hour.
* Send date/time ie date, hour, and minute.
* Send date/time ie date, hour, minute, and second.
* Send date/time ie default: Libpri will send date and hhmm only when in
NT PTMP mode to support ISDN phones.
Damien Wedhorn [Mon, 16 May 2011 21:39:33 +0000 (21:39 +0000)]
Fix up skinny hints.
Probably haven't been working for a couple of years. May still need
some more love, but they are now working, both as a hint device and
monitoring a hint. Changes centre around the long ago change
to remove the requirement for a device name in a skinny line, and
changes to the transmit_* functions.
Makes busy detection in dsp.c always allow for at least one frame (20ms) of error so that 200ms tone lengths don't get ignored by single frame error lengths.
........
Deadlock between generic CCSS agent and native ISDN CCSS.
Deadlock can occur when the generic CCSS agent is deleting duplicate CC
offers and the native ISDN CC driver is processing an incoming CC message.
The cc_core_instances container lock cannot be held when an agent or
monitor callback is invoked without the possibility of a deadlock.
* Make kill_duplicate_offers() remove the reference in cc_core_instances
outside of the container lock.
Damien Wedhorn [Sun, 15 May 2011 23:17:57 +0000 (23:17 +0000)]
Add activatesub and dialandactivate sub.
When called, activatesub first cleans up the active sub and then
handles the sub passed. dialandactivatesub first sets sub->exten
and then calls activatesub. Revise handle_offhook to utilise the
callid sent to chan_skinny. Some other minor fixes especially around
d->hookstate (which still needs some more work).
CDR's are being written immediately on caller hangup.
CDR's are being written immediately on caller hangup. The dialplan is not
able to modify it in the h exten. The h exten in the initial context is
not run before closing CDR's when the bridge is unlinked if a macro is
active and does not have an h exten.
* Make ast_bridge_call() check for an h exten in the current context and
if a macro is active then the initial context. The first h exten found is
then run before closing the CDR.
Damien Wedhorn [Fri, 13 May 2011 08:33:35 +0000 (08:33 +0000)]
Move exten used for dialing from device to subchannel.
There were some issues where if a simple switch was cancelled and a
new switch started before the first had timed out where the d->exten
would be used for both subchannels. This was bad leading to possible
invalid extensions if some digits had been entered in the abandoned
simple switch and the second one was completed before the first timed
out, or the second would be cancelled because d->exten would be set to
nothing on the time out of the first.
Fix directed group pickup feature code *8 with pickupsounds enabled
Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues.
1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked.
2). dialplan applications for directed_pickups shouldn't beep.
3). feature code for directed pickup should beep on success/failure if configured.
Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite.
Moved app_directed:pickup_do() to features:ast_do_pickup().
Functions below, all now use the new ast_do_pickup()
app_directed_pickup.c:
pickup_by_channel()
pickup_by_exten()
pickup_by_mark()
pickup_by_part()
features.c:
ast_pickup_call()
projects / thirdparty / asterisk.git / log
