Age | Commit message (Collapse) | Author |
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and it isn't found.
ASTERISK-24612 #close
Reported by: Joshua Colp
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to a non-existent extension.
ASTERISK-24716 #close
Reported by: Rusty Newton
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r431751 | file | 2015-02-14 14:19:07 -0400 (Sat, 14 Feb 2015) | 5 lines
chan_pjsip: Fix crash when CHANNEL dialplan function is invoked with pjsip argument and no type.
ASTERISK-24771 #close
Reported by: Niklas Larsson
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r431752 | file | 2015-02-14 14:20:27 -0400 (Sat, 14 Feb 2015) | 2 lines
'information' ends with an 'n'.
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A multi-asterisk box setup with direct media enabled would occasionally
crash when two re-INVITE collisions on a call leg happen in a row.
The re-INVITE logic only had one timer struct to defer the re-INVITE.
When the second collision happens the timer struct is overwritten and put
into the timer heap again. Resources for the first timer are leaked and
the heap has two positions occupied by the same timer struct. Now the
heap ordering is potentially corrupted, the timer will fire twice, and any
resources allocated for the second timer will be released twice.
* The solution is to put the collided re-INVITE into the delayed requests
queue with all the other delayed requests and cherry pick the next request
that can come off the queue when an event happens.
* Changed to put delayed BYE requests at the head of the delayed queue.
There is no sense in processing delayed UPDATEs and re-INVITEs when a BYE
has been requested.
* Made the start of a BYE request flush the delayed requests queue to
prevent a delayed request from overlapping the BYE transaction. I saw a
few cases where a delayed re-INVITE got started after the BYE transaction
started.
* Changed the delayed_request struct to use an enum instead of a string
for the request method. Cherry picking the queue is easier with an enum
than string comparisons and the compiler can warn if a switch statement
does not cover all defined enum values.
* Improved the debug output to give more information. It helps to know
which channel is involved with an endpoint. Trunks can have many channels
associated with the endpoint at the same time.
ASTERISK-24727 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4414/
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This patch adds a new feature to ARI to redirect a channel to another server,
and fixes a few bugs in PJSIP's handling of the Transfer dialplan
application/ARI redirect capability.
*New Feature*
A new operation has been added to the ARI channels resource, redirect. With
this, a channel in a Stasis application can be redirected to another endpoint
of the same underlying channel technology.
*Bug fixes*
In the process of writing this new feature, two bugs were fixed in the PJSIP
stack:
(1) The existing .transfer channel callback had the limitation that it could
only transfer channels to a SIP URI, i.e., you had to pass
'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is
still supported, it is somewhat unintuitive - particularly in a world full
of endpoints. As such, we now also support specifying the PJSIP endpoint to
transfer to.
(2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by
updating its Contact header. Alas, that resulted in the forwarding
destination set by the dialplan application/ARI resource/whatever being
rewritten with very incorrect information. Hence, we now don't bother
updating an outgoing response if it is a 302. Since this took a looong time
to find, some additional debug statements have been added to those modules
that update the Contact headers.
Review: https://reviewboard.asterisk.org/r/4316/
ASTERISK-24015 #close
Reported by: Private Name
ASTERISK-24703 #close
Reported by: Matt Jordan
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There have been a couple of times where a crash occurred in the dtls_handler
section of the code for res_pjsip. Unfortunately, in working this issue the
problem was unable to be reproduced. After looking at the backtraces and
through the code the current best guess as to why this happened might be due
to a reentrance problem and the strtok function. So, the current fix is to
convert the strtok function into the reentrant version of the function,
strtok_r.
ASTERISK-24741 #close
Reported by: Zane Conkle
Review: https://reviewboard.asterisk.org/r/4409/
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When merging the websocket timeout issue (ASTERISK-24701) an extra, almost
duplicate, check was left in the code that should not have been. This removes
it.
ASTERISK-24701 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4412/
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There are three CLI commands to stop and restart Asterisk each.
1) core stop/restart now - Hangup all calls and stop or restart Asterisk.
New channels are prevented while the shutdown request is pending.
2) core stop/restart gracefully - Stop or restart Asterisk when there are
no calls remaining in the system. New channels are prevented while the
shutdown request is pending.
3) core stop/restart when convenient - Stop or restart Asterisk when there
are no calls in the system. New calls are not prevented while the
shutdown request is pending.
ARI has made stopping/restarting Asterisk more problematic. While a
shutdown request is pending it is desirable to continue to process ARI
HTTP requests for current calls. To handle the current calls while a
shutdown request is pending, a new committed to shutdown phase is needed
so ARI applications can deal with the calls until the system is fully
committed to shutdown.
* Added a new shutdown committed phase so ARI applications can deal with
calls until the final committed to shutdown phase is reached.
* Made refuse new HTTP requests when the system has reached the final
system shutdown phase. Starting anything while the system is actively
releasing resources and unloading modules is not a good thing.
* Split the bridging framework shutdown to not cleanup the global bridging
containers when shutting down in a hurry. This is similar to how other
modules prevent crashes on rapid system shutdown.
* Moved ast_begin_shutdown(), ast_cancel_shutdown(), and
ast_shutting_down(). You should not have to include channel.h just to
access these system functions.
ASTERISK-24752 #close
Reported by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/4399/
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When a SIP device that has its registration stored in RealTime unregisters,
the entry for that device is updated with blank values, i.e., "", indicating
that it is no longer registered. Unfortunately, one of those values that is
'blanked' is the device's port. If the column type for the port is not a
string datatype (the recommended type is integer), an ODBC or database error
will be thrown. MariaDB does not coerce empty strings to a valid integer value.
This patch updates the query run from chan_sip such that it replaces the port
value with a value of '0', as opposed to a blank value. This is the value that
other database backends coerce the empty string ("") to already, and the
handling of reading a RealTime registration value from a backend already
anticipates receiving a port of '0' from the backends.
ASTERISK-24772 #close
Reported by: Richard Miller
patches:
chan_sip.diff uploaded by Richard Miller (License 5685)
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Add ast_module_shutdown_ref for use by modules that can
only be unloaded during graceful shutdown.
When REF_DEBUG is enabled:
* Add an empty ao2 object to struct ast_module.
* Allocate ao2 object when the module is loaded.
* Perform an ao2_ref in each place where mod->usecount is manipulated.
* ao2_cleanup on module unload.
ASTERISK-24479 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4141/
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When writing to a websocket if a timeout occurred the underlying socket did not
get closed/disconnected. This patch makes sure the websocket gets disconnected
on a write timeout. Also a notice is logged stating that the websocket was
disconnected.
ASTERISK-24701 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4412/
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Looking at the Super Awesome Company sample reminded me that creating hints is
just plain gruntwork. So you can now have the pjsip conifg wizard auto-create
them for you.
Specifying 'hint_exten' in the wizard will create
'exten => <hint_exten>,hint/PJSIP/<wizard_id>'
in whatever is specified for 'hint_context'.
Specifying 'hint_application' in the wizard will create
'exten => <hint_exten>,1,<hint_application>'
in whatever is specified for 'hint_context'.
The default for 'hint_context' is the endpoint's context.
There's no default for 'hint_application'. If not specified, no app is added.
There's no default for 'hint_exten'. If not specified, neither the hint itself
nor the application will be created.
Some may think this is the slippery slope to users.conf but hints are a basic
necessity for phones unlike voicemail, manager, etc that users.conf creates.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4383/
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One of the canonical reasons for hanging up a channel is because the far end
failed to answer - or because someone else answered, and we want to get rid of
this channel. This patch adds the missing value to the 'reason' query parameter
for the DELETE /channels operation.
Review: https://reviewboard.asterisk.org/r/4400
ASTERISK-24745 #close
Reported by: Ben Merrills
patches:
add_no_answer_ari_hangup_cause.diff uploaded by Ben Merrills (License 6678)
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This patch modifies the ast_odbc_find_table function such that it only performs
a lookup of the requested table if the table is not already known. Prior to
this patch, a queries would be executed against the database even if the table
was already known and cached.
Review: https://reviewboard.asterisk.org/r/4405/
ASTERISK-24742 #close
Reported by: ibercom
patches:
patch.diff uploaded by ibercom (License 6599)
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When an SDP is created for an outgoing request/response, the ICE candidates
obtained from the RTP instance are currently leaked. This causes the ao2
container that holds the candidates to never properly be reclaimed when the
RTP instance is destroyed.
This patch properly decrements the ICE candidates' container if it is
successfully obtained.
ASTERISK-24769 #close
Reported by: Matt Jordan
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In this collection of small patches to prevent
Valgrind errors are: fixes for reference leaks
in config hooks, evaluating a parameter beyond
bounds, and accessing a structure after a lock
where it could have been already free'd.
Review: https://reviewboard.asterisk.org/r/4407/
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The act of defining wizards for an object type in sorcery.conf will
create a minimal object type. This can cause a problem when a module
has multiple sorcery instances (which all get the wizards from sorcery.conf
applied) but the sorcery instances do not all contain full information
about the object types. Upon loading errors will occur stating that
the objects can not be created. This is confusing and is actually
perfectly fine.
This change makes it so that only object types which have been fully
defined will be loaded.
ASTERISK-24748 #close
Reported by: Joshua Colp
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The res_format_attr_h264 module currently incorrectly attempts to
copy SPS and PPS information from the wrong attribute. This change
fixes that.
ASTERISK-24616 #close
Reported by: Yura Kocyuba
Review: https://reviewboard.asterisk.org/r/4392/
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When the app_agent_pool module initially loads there is a race condition
between the thread loading agents.conf and the device state internal
processing thread. If the device state internal processing thread handles
the agent creation state updates before the thread that loaded agents.conf
registers the device state provider callback then the cached agent state
is "Invalid". When a consumer module like app_queue asks for the agent state
it gets the cached "Invalid" state instead of the real state from the provider.
* Moved loading the agents.conf configuration to the last thing setup by
app_agent_pool in load_module(). Now the device state provider callback
is registered before the config is loaded so the agent creation state
updates are guaranteed to get the initial device state.
* Removed some now redundant config cleanup on error in load_config().
* Added lock protection when accessing the device state in
agent_pvt_devstate_get() and eliminated the RAII_VAR() usage.
ASTERISK-24737 #close
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4390/
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When Asterisk attempts to send SIP outbound publish information and no response
is ever received (no 200 okay, 412, 423) the system eventually crashes. A
response is never received because the system Asterisk is attempting to send
publish information to is not available. The underlying pjsip framework attempts
to send publish information. After several attempts it calls back into the
Asterisk outbound publish code. At this point if the "client->queue" is empty
Asterisk attempts to schedule a refresh which utilizes "rdata" and since no
response was received the given "rdata" struture is NULL. Attempting to
dereference a NULL object of course results in a crash.
The fix here removes the dependency on rdata for schedule_publish_refresh.
Instead param->expiration is now passed to it as this is set to -1 if no
response is received. Also added a notification when no response is received.
ASTERISK-24635 #close
Reported by: Marco Paland
Review: https://reviewboard.asterisk.org/r/4384/
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Added a new config property [servername] to the http.conf file; updated the http server to use the new property when sending responses, for showing http status through the CLI and when reporting status through the 'httpstatus' webpage. In this version, [servername] is uncommented by default.
ASTERISK-24316 #close
Reported By: Andrew Nagy
Review: https://reviewboard.asterisk.org/r/4374/
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These memory leaks were found and fixed by John Hardin. I'm just
committing them for him.
ASTERISK-24736 #close
Reported by Mark Michelson
Review: https://reviewboard.asterisk.org/r/4389
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When swapping a Local channel in place of one already
in a bridge (to complete a bridge attended transfer),
the channel that was swapped out can actually be hung
up before the stasis bridge push callback executes on
the independant transfer thread. This results in the
stasis app loop dropping out and removing the control
that has the the app name which the local replacement
channel needs so it can re-enter stasis.
To avoid this race condition a new push_peek callback
has been added, and called from the ast_bridge_impart
thread before it launches the independant thread that
will complete the transfer. Now the stasis push_peek
callback can copy the stasis app name before the swap
channel can hang up.
ASTERISK-24649
Review: https://reviewboard.asterisk.org/r/4382/
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Due to an inversion error, setting 100rel=no would not actually
change the current value of the setting (which defaulted to "yes").
With this fix, the inversion is corrected.
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RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific
scenarios when we are required to use SIPS URIs in Contact
headers. Asterisk's non-compliance with this could actually
cause calls to get dropped when communicating with clients
that are strict about checking the Contact header.
Both of the SIP stacks in Asterisk suffered from this issue.
This changeset corrects the behavior in res_pjsip/chan_pjsip.c
Review: https://reviewboard.asterisk.org/r/4345
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RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific
scenarios when we are required to use SIPS URIs in Contact
headers. Asterisk's non-compliance with this could actually
cause calls to get dropped when communicating with clients
that are strict about checking the Contact header.
Both of the SIP stacks in Asterisk suffered from this issue.
This changeset corrects the behavior in chan_sip.
ASTERISK-24646 #close
Reported by Stephan Eisvogel
Review: https://reviewboard.asterisk.org/r/4346
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Reduce log clutter by changing the "Watcher for hint %s (removed|deactivated)"
message from WARNING to VERBOSE/2.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4387/
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A recent security fix for OpenSSL broke DTLS negotiation for many
applications. This was caused by read ahead not being enabled when it
should be. While a commit has gone into OpenSSL to force read ahead
on for DTLS it may take some time for a release to be made and the
change to be present in distributions (if at all). As enabling read
ahead is a simple one line change this commit does that and fixes
the issue.
ASTERISK-24711 #close
Reported by: Jared Biel
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SIP requests that offered codecs incompatible with configured values
could result in the allocation of RTP and RTCP ports that would not get
reclaimed later.
ASTERISK-24666 #close
Reported by Y Ateya
Review: https://reviewboard.asterisk.org/r/4323
AST-2015-001
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r431297 | mmichelson | 2015-01-28 11:05:26 -0600 (Wed, 28 Jan 2015) | 17 lines
Mitigate possible HTTP injection attacks using CURL() function in Asterisk.
CVE-2014-8150 disclosed a vulnerability in libcURL where HTTP request injection
can be performed given properly-crafted URLs.
Since Asterisk makes use of libcURL, and it is possible that users of Asterisk may
get cURL URLs from user input or remote sources, we have made a patch to Asterisk
to prevent such HTTP injection attacks from originating from Asterisk.
ASTERISK-24676 #close
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/4364
AST-2015-002
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r431298 | mmichelson | 2015-01-28 11:12:49 -0600 (Wed, 28 Jan 2015) | 3 lines
Fix compilation error from previous patch.
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In r419044, we changed how formats were handled, but the return value
of the format_parse_sdp_fmtp functions in res_format_attr_opus and
res_format_attr_silk were not updated, causing calls to fail. Ran
into this when getting codec_opus working with Asterisk 13.
Once the return value was corrected, we were crashing in opus_getjoint
because of NULL format attributes. I've fixed this as well in this
patch.
Review: https://reviewboard.asterisk.org/r/4371/
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Performing a CLI "module reload" command when there are new pjsip.conf
registration objects defined frequently failed to load them correctly.
What happens is a race condition between res_pjsip pushing its reload into
an asynchronous task processor task and the thread that does the rest of
the reloads when it gets to reloading the res_pjsip_outbound_registration
module. A similar race condition happens between a reload and the CLI/AMI
show registrations commands. The reload updates the current_states
container and the CLI/AMI commands call get_registrations() which builds a
new current_states container.
* Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous()
instead of ast_sip_push_task() to eliminate two threads processing config
reloads at the same time.
* Made get_registrations() not replace the global current_states container
so the CLI/AMI show registrations command cannot interfere with reloading.
You could never add/remove objects in the container without the
possibility of the container being replaced out from under you by
get_registrations().
* Added a registration loaded sorcery instance observer to purge any dead
registration objects since get_registrations() cannot do this job anymore.
The struct ast_sorcery_instance_observer callbacks must be used because
the callback happens inline with the load process. The struct
ast_sorcery_observer callbacks are pushed to a different thread.
* Added some global current_states NULL pointer checks in case the
container disappears because of unload_module().
* Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded
callbacks guaranteed to be called before any struct
ast_sorcery_observer.loaded callbacks will be called.
* Moved the check for non-reloadable objects to before the sorcery
instance loading callbacks happen to short circuit unnecessary work.
Previously with non-reloadable objects, the sorcery instance
loading/loaded callbacks would always happen, the individual wizard
loading/loaded would be prevented, and the non-reloadable type logging
message would be logged for each associated wizard.
ASTERISK-24729 #close
Review: https://reviewboard.asterisk.org/r/4381/
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While running through some scenarios using chan_sip and tcp a problem would
occur that resulted in a flood of bad file descriptor messages on the cli:
tcptls.c:712 ast_tcptls_server_root: Accept failed: Bad file descriptor
The message is received because the underlying socket has been closed, so is
valid. This is probably happening because unloading of chan_sip is not atomic.
That however is outside the scope of this patch. This patch simply stops the
logging of multiple occurrences of that message.
ASTERISK-24728 #close
Reported by: Thomas Thompson
Review: https://reviewboard.asterisk.org/r/4380/
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Prior to this patch, ModuleLoad would respond with an error indicating that
the requested module wasn't found in spite of finding and reloading the
module.
Review: https://reviewboard.asterisk.org/r/4373/
ASTERISK-24721 #close
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When refreshing (with a small expiration) a registration that was sent to
chan_sip the nonce would be considered stale and reject the registration.
What was happening was that the initial registration's "dialog" still existed
in the dialogs container and upon refresh the dialog match algorithm would
choose that as the "dialog" instead of the newly created one. This occurred
because the algorithm did not check to see if the from tag matched if
authentication info was available after the 401. So, it ended up assuming
the original "dialog" was a match and stopped the search. The old "dialog"
of course had an old nonce, thus the stale nonce message.
This fix attempts to leave the original functionality alone except in the case
of a REGISTER. If a REGISTER is received if searches for an existing "dialog"
matching only on the callid. If the expires value is low enough it will reuse
dialog that is there, otherwise it will create a new one.
ASTERISK-24715 #close
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/4367/
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Due to the original patch causing memory corruptions it was removed until the
problem could be resolved. This patch is the original patch plus some added
locking around stasis router subcription that was needed to avoid the memory
corruption.
Description of the original problem and patch (still applicable):
The res_pjsip module was previously unloadable. With this patch it can now
be unloaded.
This patch is based off the original patch on the issue (listed below) by Corey
Farrell with a few modifications. Namely, removed a few changes not required to
make the module unloadable and also fixed a bug that would cause asterisk to
crash on unloading.
This patch is the first step (should hopefully be followed by another/others at
some point) in allowing res_pjsip and the modules that depend on it to be
unloadable. At this time, res_pjsip and some of the modules that depend on
res_pjsip cannot be unloaded without causing problems of some sort.
The goal of this patch is to get res_pjsip and only res_pjsip to be able to
unload successfully and/or shutdown without incident (crashes, leaks, etc...).
Other dependent modules may still cause problems on unload.
Basically made sure, with the patch applied, that res_pjsip (with no other
dependent modules loaded) could be succesfully unloaded and Asterisk could
shutdown without any leaks or crashes that pertained directly to res_pjsip.
ASTERISK-24485 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4363/
patches:
pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909)
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recording channel.
Starting and stopping conference recording more than once causes the
recording channels to be leaked. For v13 the channels also show up in the
CLI "core show channels" output.
* Reworked and simplified the recording channel code to use
ast_bridge_impart() instead of managing the recording thread in the
ConfBridge code. The recording channel's ref handling easily falls into
place and other off nominal code paths get handled better as a result.
ASTERISK-24719 #close
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/4368/
Review: https://reviewboard.asterisk.org/r/4369/
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direct media.
This change fixes two issues:
1. During a swap operation bridging added the new channel before having the swap channel
leave. This was not handled in bridge_native_rtp and could result in a channel not getting
reinvited back to Asterisk. After this change the swap channel will leave first and the
new channel will then join.
2. If a re-invite was received after a session had been established any upstream elements
(such as bridge_native_rtp) were not notified that they may want to re-evaluate things.
After this change an UPDATE_RTP_PEER control frame is queued when this situation occurs
and upstream can react.
AST-1524 #close
Review: https://reviewboard.asterisk.org/r/4378/
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This patch improves the documentation of ARI on the wiki. Specifically, it
addresses the following:
* Allowed values and allowed ranges weren't documented. This was particularly
frustrating, as Asterisk would reject query parameters with disallowed values
- but we didn't tell anyone what the allowed values were.
* The /play/id operation on /channels and /bridges failed to document all of
the added media resource types.
* Documentation for creating a channel into a Stasis application failed to
note when it occurred, and that creating a channel into Stasis conflicts with
creating a channel into the dialplan.
* Some other minor tweaks in the mustache templates, including italicizing the
parameter type, putting the default value on its own sub-bullet, and some
other nicities.
Review: https://reviewboard.asterisk.org/r/4351
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When issuing a 'confbridge list XXXX' CLI command, the resulting output no
longer displays the menu associated with a ConfBridge participant.
The issue was caused by ASTERISK-22760. When that patch was done, it removed
the copying of the menu name associated with the user from the actual user
profile.
This patch fixes the issue by copying the menu name over to the user profile
when the menu hooks are applied to the user. Since that function now does a
little bit more than just apply the hooks, the name of the function has been
changed to cover the copying of the menu name over as well.
In addition, there is a disparity between the menu name length as it is stored
on the conf_menu structure and the confbridge_user structure; this patch makes
the lengths match so that a strcpy can be used.
Review: https://reviewboard.asterisk.org/r/4372/
ASTERISK-24723 #close
Reported by: Steve Pitts
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There is currently a race condition when unloading the res_parking
module. Depending on the will of the universe the subscription
invocation may occur AFTER the module is unloaded. This is because
the module does NOT use stasis_unsubscribe_and_join when terminating
the subscription. It merely uses stasis_unsubscribe.
This change makes it use stasis_unsubscribe_and_join which is documented
for usage in this exact scenario.
AST-1520 #close
Review: https://reviewboard.asterisk.org/r/4375/
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This patch addresses compilation errors on OS X. It's been a while, so
there's quite a few things.
* Fixed __attribute__ decls in route.h to be portable.
* Fixed htonll and ntohll to work when they are defined as macros.
* Replaced sem_t usage with our ast_sem wrapper.
* Added ast_sem_timedwait to our ast_sem wrapper.
* Fixed some GCC 4.9 warnings using sig*set() functions.
* Fixed some format strings for portability.
* Fixed compilation issues with res_timing_kqueue (although tests still fail
on OS X).
* Fixed menuconfig /sbin/launchd detection, which disables res_timing_kqueue
on OS X).
ASTERISK-24539 #close
Reported by: George Joseph
ASTERISK-24544 #close
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/4327/
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When a crash was fixed due to usage of the REALTIME function in r423003, a
regression was introduced into ast_update2_realtime where the update fields
passed to the function would be skipped and the lookup field processed twice.
The use of this function is a bit interesting: A variable argument list is
used with two sentinel values - the first marks the end of the lookup
fields/values; the second marks the end of the update fields/values.
Unfortunately, ast_update2_realtime parses over the lookup fields twice, as
opposed to parsing over the update fields. This causes the lookups to succeed,
but the updates itself to have no effect.
Note that the most common instance of this problem occurred in app_voicemail
during the updating of a mailbox password.
Thanks to the issue reporter, Paddy Grice, for pointing out the problem.
Review: https://reviewboard.asterisk.org/r/4356/
ASTERISK-24231
ASTERISK-24626 #close
Reported by: Paddy Grice
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Channel names should be different from other channels in the system while
the channel exists.
* Use a sequence number for CBRec channels instead of a random number
because the same random number could be picked again for the next CBRec
channel.
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Because there is sometimes no sence to any whitespace.
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Fixed memory leaks that were found in Asterisk.
ASTERISK-24693 #close
Reported by: Kevin Harwell
Review: https://reviewboard.asterisk.org/r/4347/
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