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
Jonathan Rose [Fri, 20 May 2011 18:12:21 +0000 (18:12 +0000)]
Fixes an imapfolder related crash
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.
Richard Mudgett [Fri, 20 May 2011 16:43:02 +0000 (16:43 +0000)]
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.
Richard Mudgett [Fri, 20 May 2011 15:48:25 +0000 (15:48 +0000)]
Crash when using directed pickup applications.
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.
Jonathan Rose [Fri, 20 May 2011 13:28:24 +0000 (13:28 +0000)]
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.
Terry Wilson [Thu, 19 May 2011 23:28:13 +0000 (23:28 +0000)]
Revert part of a change to the bridging API code
The capabilities used in the bridging API are very different than the
ones used for formats. When the conversion was made expanding the bit
width of codecs, the bridging code was accidentally accosted in ways
that it didn't deserve.
Richard Mudgett [Thu, 19 May 2011 16:50:48 +0000 (16:50 +0000)]
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.
........
................
Terry Wilson [Wed, 18 May 2011 20:22:36 +0000 (20:22 +0000)]
Unbreak the storing of registrations for restart
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.
Jonathan Rose [Mon, 16 May 2011 21:00:55 +0000 (21:00 +0000)]
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.
Richard Mudgett [Mon, 16 May 2011 20:33:37 +0000 (20:33 +0000)]
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.
Richard Mudgett [Fri, 13 May 2011 16:28:26 +0000 (16:28 +0000)]
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.
Alec L Davis [Thu, 12 May 2011 22:52:08 +0000 (22:52 +0000)]
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()
Several situations in the code could lead to peers or sip_pvt references
being leaked. This would cause RTP ports to never be destroyed (leading
to exhaustion of all available RTP ports) and memory leaks.
The original patch for this issue from rgagnon was the result of an
obscene amount of testing and hard work, for which I am very grateful. I
did some cleanup and added a few additional refcount fixes that I found.
Richard Mudgett [Tue, 10 May 2011 23:41:08 +0000 (23:41 +0000)]
Unable to pickup DAHDI/PRI call because call state is reported as DIALING.
The channel state is not updated to RINGING when an ALERTING message is
received. Regression caused when sig_pri.c (also sig_ss7.c) extracted
from chan_dahdi.c.
* Added missing channel state update to RINGING when the
AST_CONTROL_RINGING frame is queued for ISDN and SS7.
Richard Mudgett [Mon, 9 May 2011 23:15:32 +0000 (23:15 +0000)]
Remove references to res_features and its export file.
The contents of res/res_features.c was moved to into main/features.c
awhile ago. There is no longer any need for the res/Makefile to reference
res_features or the res_features linker exports file to exist.
Don't offer video to directmedia callee unless caller offered it as well
Make sure that when directmedia is enabled, that video is not offered to the
callee even if it supports it. p->vrtp will not exist since the caller didn't
offer video.
Richard Mudgett [Mon, 9 May 2011 19:07:01 +0000 (19:07 +0000)]
Hangup extension executed twice.
When a user hangs up a call, in certain circumstances, the hangup
extension can end up being executed twice:
1) If a call is bridged and the 'h' extension executes the Hangup
application, then the 'h' extension will be executed twice.
2) If a call is bridged within a macro (Dial or Queue), it has its own 'h'
extension, the main context also has an 'h' extension, and the macro 'h'
extension executes the Hangup application, then both 'h' extensions will
be executed.
* Revert originally commited fix for #16106 and just set
AST_FLAG_BRIDGE_HANGUP_RUN unconditionally in ast_bridge_call(). The
bridge code just executed an 'h' extension so the main PBX loop does not
need to execute one as well.
Richard Mudgett [Mon, 9 May 2011 16:57:18 +0000 (16:57 +0000)]
Don't get early media for ISDN on outgoing calls.
It looks to be a long-standing misinterpretation of the progress indicator
ie values:
1 - Call is not end-to-end ISDN; further call progress information may be
available in-band.
8 - In-band information or an appropriate pattern is now available.
Only value 8 is handled by chan_dahdi/sig_pri. The 1 value is not handled
as early media probably because the meaning of the second half of it's
description was overlooked.
* Test to see if either PRI_PROG_CALL_NOT_E2E_ISDN(1) or
PRI_PROG_INBAND_AVAILABLE(8) bits are set to open the media path.
No inband progress on PRI_EVENT_RINGING even if inband flag set.
My ISDN-PRI provider sends an ALERTING with "Inband information or
appropriate pattern now available", but Asterisk only generates and passes
the RING to the SIP extension, not the inband message. Unfortunately, the
inband message is not a ringback tone but a prompt that says the number is
not in service. The SIP extension then hears two rings and the call is
hungup which confuses the caller.
* Post an AST_CONTROL_PROGRESS as well as opening the media path if inband
audio is indicated with an ALERTING message.
Jonathan Rose [Mon, 9 May 2011 14:18:14 +0000 (14:18 +0000)]
Documenting an observed behavior of features in features.conf. Since parkinglots use an
integer for the parkinglot extensions, leading zeros specified in the configuration file
are ignored.
Russell Bryant [Fri, 6 May 2011 19:46:49 +0000 (19:46 +0000)]
chan_sip: fix a deadlock in check_rtp_timeout.
Don't block doing silly deadlock avoidance. Just return and try again later.
The funciton gets called often enough that it's fine. Also, this change was
already made in trunk.
Don't start an autoservice in pbx_lua if pbx_lua already started one and don't
stop one if we didn't start one. Also start and stop the autoservice when
transferring control from and to the pbx.
Richard Mudgett [Fri, 6 May 2011 16:19:18 +0000 (16:19 +0000)]
Fix SIP connected line updates.
This patch fixes a couple SIP connected line update problems:
1) The connected line needs to be updated when the initial INVITE is sent
if there is a peer callerid configured. Previously, the connected line
information did not get reported until the call was connected so SIP could
not report connected line information in ringing or progress messages.
2) The connected line should not be updated on initial connect if there is
no connected line information. Previously, all it did was wipe out any
default preset CONNECTEDLINE information set by the dialplan with empty
strings.
This part of the change for r315596 was incorrect. No bridge occurs
when doing a roundrobin dial and no one answers, so this code shouldn't
have been removed.
........
................
Russell Bryant [Thu, 5 May 2011 22:53:45 +0000 (22:53 +0000)]
Fix some consistency issues with jitterbuffer config.
Store the defaults noted in the sample config files in the jitterbuffer config
data structure. This makes the CLI commands that output these settings show
the right thing. Also only show the settings that are relevant in the settings
CLI commands, based on which jitterbuffer is selected and whether it's enabled.
Jonathan Rose [Thu, 5 May 2011 19:09:13 +0000 (19:09 +0000)]
Resolves a deadlock that occurs during sip_new
This is based on an uncommitted patch by jpeeler for the issue. Instead of
relocking and then unlocking the channel though, we keep the lock on the channel
until we are finished doing what we need to the channel.
Set SO_KEEPALIVE on SIP TCP sockets so that they eventually go away when a peer
abruptly disappears. This mostly occurs after a successful registration.
(closes issue #17544)
Reported by: marcelloceschia
Patches:
(modified) tcptls.patch uploaded by st (license 907)
Sean Bright [Thu, 5 May 2011 02:30:45 +0000 (02:30 +0000)]
Use the correct HTTP method when generating our digest, otherwise we always fail.
When calculating the 'A2' portion of our digest for verification, we need the
HTTP method that is currently in use. Unfortunately our mapping function was
incorrect, resulting in invalid hashes being generated and, in turn, failures
in authentication.
Fixes session-timers=refuse not being enforced for *caller*
During handle_request_invite, the session timer mode was retrieved from
a cached variable. This patch forces a peer lookup of the session timer
mode in the case of an incoming invite.
(closes issue #18804)
Reported by: wdoekes
Patches:
issue18804_session_timer_refuse_caller.patch uploaded by wdoekes (license 717)
issue_18804_v2.diff uploaded by dvossel (license 671)
........
Richard Mudgett [Tue, 3 May 2011 19:18:30 +0000 (19:18 +0000)]
The dahdi_hangup() call does not clean up the channel fully.
After dahdi_hangup() has supposedly hungup an ISDN channel there is still
traffic on the S0-bus because the channel was not cleaned up fully.
Shuffled the hangup code to include some missing cleanup. Also fixed some
code formatting in the area. I think the primary missing clean up code
was the call to tone_zone_play_tone() to turn off any active tones on the
channel.
David Vossel [Tue, 3 May 2011 18:59:06 +0000 (18:59 +0000)]
Never put the Require: timer header in an Invite.
This has already been discussed and should have been resolved earlier. View
revsion 285565's log for more information about why it is important to not
put timer in the Require header.
This change optimizes the free_via() function and removes some redundant null
checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using
the port specified in the Via header for routing responses (even when maddr is
not set). Also the htons() function is now used when setting the port.
Additional documentation comments have been added in various places to make the
logic in the code clearer.
projects / thirdparty / asterisk.git / log
