Age | Commit message (Collapse) | Author |
|
The 'pjsip_get_dest_info' function is used to determine if the signaling transport
of the dialog is secure or not. This function was added in PJSIP 2.3 and does not
exist in earlier versions.
This configure check allows Asterisk to build and run with older versions at the
loss of the 'secure' argument for the PJSIP CHANNEL dialplan function. Usage of
this argument will require upgrading to PJSIP 2.3.
ASTERISK-24665 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4329/
........
Merged revisions 430546 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* Reverted the change to astman_send_listack() to not use the listflag
parameter and always set the value to "Start" so the start capitalization
is consistent. Unfortunately changing the case of a returned value is not
a backward compatible change so for now FAXSessions is going to have to
remain inconsistent with all of the other AMI list actions.
* Reverted the minor protocol error fix in action_getconfig() when no
requested categories are found. Each line needs to be formatted as
"Header: text".
Caught by the testsuite.
ASTERISK-24049
........
Merged revisions 430528 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The sample config was missing the configuration options for DTMF attended
transfer completion scenarios. The configuration options 'atxferabort',
'atxfercomplete', 'atxferthreeway', and 'atxferswap' are now documented in the
appropriate configuration file.
ASTERISK-24678 #close
Reported by: Niklas Larsson
patches:
features.conf.sample.diff uploaded by Niklas Larsson (License 5068)
........
Merged revisions 430526 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
instead.
ASTERISK-24049
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The security event log uses a dynamic log level (SECURITY) that is registered
with the Asterisk logging core. Unfortunately, the syslog would ignore log
statements that had a dynamic log level associated with them. Because the
syslog cannot handle ad hoc dynamic log levels, this patch treats any dynamic
log entries sent to the syslog as logs with a level of NOTICE.
ASTERISK-20744 #close
Reported by: Michael Keuter
Tested by: Michael L. Young, Jacek Konieczny
patches:
asterisk-20744-syslog-dynamic-logging_trunk.diff uploaded by Michael L. Young (license 5026)
........
Merged revisions 430506 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 430507 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
When the channel datastore associated with the usage of CURLOPT on a specific
channel is freed, the underlying structure holding the list of options is not
disposed of. This patch properly frees the structure in the datastore .destroy
callback.
ASTERISK-24672 #close
Reported by: Kristian Hogh
patches:
func_curl-memory-leak.diff uploaded by Kristian Hogh (License 6639)
........
Merged revisions 430487 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 430488 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
General improvements to SIP to PJSIP conversion utility:
1) track default section of input file to allow parsing
an include file that doesn't specify a [section]
2) informatively handle case of assignment without [section]
3) correctly handle getting sections from included files
- [section]'s are inherited by included file
4) provide null string as default transport bind ip
5) gracefully handle missing portions of registration string
6) denote steps of operation during conversion and confirm
top level files as a convenience
ASTERISK-24474 #close
Review: https://reviewboard.asterisk.org/r/4280/
Reported by: John Kiniston
........
Merged revisions 430469 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
When app_bridge grabs a channel and puts it into
a bridge, the channel should then continue where
it left off in the dialplan after the bridge has
ended. Although it stores the current dialplan
location as an after bridge goto on the channel,
it was executing the same priority again instead
of going to the next priority. By swapping the
"specific" version of bridge_set_after_goto with
bridge_set_after_go_on, the next priority in the
dialplan is executed instead.
ASTERISK-24637 #close
Review: https://reviewboard.asterisk.org/r/4322/
Reported by: John Bigelow
........
Merged revisions 430467 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Follow-up issue to -r430435 from reviewboard review.
ASTERISK-24049
Review: https://reviewboard.asterisk.org/r/4315/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* Made the following AMI actions use list API calls for consistency:
Agents
BridgeInfo
BridgeList
BridgeTechnologyList
ConfbridgeLIst
ConfbridgeLIstRooms
CoreShowChannels
DAHDIShowChannels
DBGet
DeviceStateList
ExtensionStateList
FAXSessions
Hangup
IAXpeerlist
IAXpeers
IAXregistry
MeetmeList
MeetmeListRooms
MWIGet
ParkedCalls
Parkinglots
PJSIPShowEndpoint
PJSIPShowEndpoints
PJSIPShowRegistrationsInbound
PJSIPShowRegistrationsOutbound
PJSIPShowResourceLists
PJSIPShowSubscriptionsInbound
PJSIPShowSubscriptionsOutbound
PresenceStateList
PRIShowSpans
QueueStatus
QueueSummary
ShowDialPlan
SIPpeers
SIPpeerstatus
SIPshowregistry
SKINNYdevices
SKINNYlines
Status
VoicemailUsersList
* Incremented the AMI version to 2.7.0.
* Changed astman_send_listack() to not use the listflag parameter and
always set the value to "Start" so the start capitalization is consistent.
i.e., The FAXSessions used "Start" while the rest of the system used
"start". The corresponding complete event always used "Complete".
* Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the
AMI ActionID for all of its list events.
* Fixed off-nominal AMI protocol error in manager_bridge_info(),
manager_parking_status_single_lot(), and
manager_parking_status_all_lots(). Use of astman_send_error() after
responding to the original AMI action request violates the action response
pattern by sending two responses.
* Fixed minor protocol error in action_getconfig() when no requested
categories are found. Each line needs to be formatted as "Header: text".
* Fixed off-nominal memory leak in manager_build_parked_call_string().
* Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail().
ASTERISK-24049 #close
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/4315/
........
Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This change makes the T.38 negotiation timeout configurable via
't38timeout' in res_fax.conf or FAXOPT(t38timeout). It was previously
hard coded to be 5000 milliseconds.
This change also handles T.38 switch failures by aborting the fax since
in the case where this can happen, both sides have agreed to switch to
T.38 and Asterisk is unable to do so.
Review: https://reviewboard.asterisk.org/r/4320/
........
Merged revisions 430415 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 430416 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
If you do a 'core (shutdown|restart) graceful' persistent subscriptions won't
survive. If you do a 'core (shutdown|restart) now' or asterisk terminates for
some reason, they do. Here's why...
When asterisk shuts down gracefully, it sends a 'NOTIFY/terminated' to
subscribers for each subscription. This not only tells the subscribers that the
dialog/state machine is done, it also frees the last reference to the
subscription tree which causes the persistent subscription to get deleted from
astdb. When asterisk restarts, nothing's left. Just preventing the delete from
astdb doesn't work because we already told the subscriber to terminate the
dialog so we can't restart it even if it was still in astdb. Everything works
OK if asterisk terminates unexpectedly because we never send the 'terminated'
message so on restart, the subscription is still in astdb and the subscriber is
none the wiser.
This patch suppresses the sending of 'NOTIFY/terminated' on shutdown for
persistent connections.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4318/
........
Merged revisions 430397 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430398 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Every time a registration started, sip_outbound_registration_response_cb bumps
the ref count on client_state then pushes a handle_registration_response task.
handle_registration_response never unreffed it though. So every time a
registration goes out, the ref count goes up by one.
This patch adds the unreffs to handle_registration_response.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4303/
........
Merged revisions 430395 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
There are 2 issues with reloading registrations...
1. The 'can_reuse_registration' test wasn't considering the intervals or
expiration in its determination of whether a registration changed or not so if
you changed any of the intervals or the expiration and reloaded, the object
would get reloaded but the actual timers wouldn't change.
can_reuse_registration now does a sorcery diff on the old and new objects
instead of discretely testing certain fields. Now if you change expiration for
instance, and reload, the timer is updated and re-registration will occur on the
new value.
2. If you mung up your password on an outbound registration you get a permanent
failure. If you fix the password (on the outbound_auth object) and reload,
nothing tells outbound_registration to try again because the registration itself
didn't change. This patch adds an observer on the "auth" object type and if any
auth changes, existing registration states are searched and those in a
REJECTED_PERMANENT state are retried.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4304/
........
Merged revisions 430373 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
When the AMI Redirect action is used with a channel bridged inside
Stasis() and not running a pbx, the channel is hung up instead of
proceeding to the desired location in dialplan. This change allows
such channels to be Redirected properly by detecting the operation
used by Redirect (ASYNCGOTO) and using the code already established
for functionality of the ARI channel continue operation.
ASTERISK-24591 #close
Review: https://reviewboard.asterisk.org/r/4271/
........
Merged revisions 430355 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430356 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
With this patch, the following two ARI commands
POST /channels
POST /channels/{id}/continue
Accept a new parameter, label, that can be used to continue to or originate
to a priority label in the dialplan.
Because this is adding a new parameter to ARI commands, the API version of
ARI has been bumped from 1.6.0 to 1.7.0.
This patch comes courtesy of Nir Simionovich from Greenfield Tech. Thanks!
ASTERISK-24412 #close
Reported by Nir Simionovich
Review: https://reviewboard.asterisk.org/r/4285
........
Merged revisions 430337 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The 'new_subscribe: Extension <> does not exist or has no associated hint'
is a config issue and doesn't need to clutter up logs with warnings.
Changed to notice.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4307/
........
Merged revisions 430319 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The "MWI Subscription failed" message means the client is trying to subscribe
to a mailbox that doesn't exist. There's no need to clutter up logs with
warnings for a client misconfiguration so I changed it to a notice.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4306/
........
Merged revisions 430317 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
I guess nobody uses templates with AST_CONFIG because today if you have a
context that inherits from a template and you call AST_CONFIG on the context,
you'll get the value from the template even if you've overridden it in the
context. This is because AST_CONFIG only gets the first occurrence which is
always from the template.
This patch adds an optional 'index' parameter to AST_CONFIG which lets you
specify the exact occurrence to retrieve, or '-1' to retrieve the last.
The default behavior is the current behavior.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4313/
........
Merged revisions 430315 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This fix has two parts:
* Corrected an error message to properly state that external_replaces is an extension. The
error message also prints what dialplan context the external_replaces extension was being
looked for in.
* Corrected the printing of the Replaces: header in an INVITE request. We were duplicating
"Replaces: " in the header.
ASTERISK-24376 #close
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/4296
........
Merged revisions 430313 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Let's say you have a template T with variable VAR1 = ON and you have a
context C(T) that doesn't specify VAR1. If you read C, the effective value
of VAR1 is ON. Now you change T VAR1 to OFF and call
ast_config_text_file_save. The current behavior is that the file gets
re-written with T/VAR1=OFF but C/VAR1=ON is added. Personally, I think this
is a bug. It's preserving the effective state of C even though I didn't
specify C/VAR1 in th first place. I believe the behavior should be that if
I didn't specify C/VAR1 originally, then the effective value of C/VAR1 should
continue to follow the inherited state. Now, if I DID explicitly specify
C/VAR1, the it should be preserved even if the template changes.
Even though I think the existing behavior is a bug, it's been that way forever
so I'm not changing it. Instead, I've created ast_config_text_file_save2()
that takes a bitmask of flags, one of which is to preserve the effective context
(the current behavior). The original ast_config_text_file_save calls *2 with
the preserve flag. If you want the new behavior, call *2 directly without a
flag.
I've also updated Manager UpdateConfig with a new parameter
'PreserveEffectiveContext' whose default is 'yes'. If you want the new behavior
with UpdateConfig, set 'PreserveEffectiveContext: no'.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4297/
........
Merged revisions 430295 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 430274 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
contrib/ast-db-manage: Correct down_revision path for user_eq_phone
When the user_eq_phone patch was backported to 13, it referenced the downward
revision that the PJSIP optimistic encryption option also references. This
creates a multi-path upgrade Exception when generating the SQL files.
This patch corrects this in the 13 branch. Note that trunk, which already
contained both of these features, is unaffected by this problem.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
When res_pjsip loads and an endpoint auto-subscribes a mailbox for mwi,
if a contact hasn't registered yet, res_pjsip_mwi spits out a warning.
This is a perfectly normal situation though and doesn't require something
as serious as a warning. It's also self correcting. The device will start
getting mwi as soon as it registers.
This patch changes the warning to a notice.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4314/
........
Merged revisions 430227 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Change the "Locally bridged"/"Remotely bridged" messages from dbg/2 to verb/4.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4300/
........
Merged revisions 430225 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The current behavior of 'pjsip send unregister' is to send the unregister
(REGISTER with 0 exp) but let the next scheduled register proceed normally.
I don't think that's a good idea. If you unregister, it should stay
unregistered until you decide to start registrations again. So this patch
just adds a cancel_registration call to the current unregister_task to
cancel the timer.
Of course, now you need a way to start registration again so I've added
a 'pjsip send register' command that unregisters and cancels any existing
registration (the same as send unregister), then sends an immediate
registration and starts the timer back up again.
Both changes also ripple to AMI. There's a new PJSIPRegister command.
There's no harm in calling either command repeatedly. They don't care
about the actual state.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4301/
........
Merged revisions 430223 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
For some reason I was using a hash container instead of a list to gather the
contacts for 'pjsip list/show contacts' so even though I had a sort function,
the output wasn't sorted. This patch just changes the hash container to a
list container and the contacts now appear sorted in the CLI.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4305/
........
Merged revisions 430221 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
A blond transfer to a failed destination, when followed
by a recall attempt, lead to a leak of the reference to
the destination channel. In addition to correcting the
regression on the previous attempt (r429826) this fixes
the leak and two additional reference leaks on failures
of bridge_import.
ASTERISK-24513 #close
Review: https://reviewboard.asterisk.org/r/4302/
........
Merged revisions 430199 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 430200 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 430181 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The PJSIP_AOR dialplan function allows inspection of configured AORs including
what contacts are currently bound to them.
The PJSIP_CONTACT dialplan function allows inspection of contacts in existence.
These can include both externally added (by way of registration) or permanent
ones.
ASTERISK-24341
Reported by: xrobau
Review: https://reviewboard.asterisk.org/r/4308/
........
Merged revisions 430179 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430180 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
In r428708 additional codecs were added including
a payload type of 128 which is outside of nominal
range of 0-127. This change moves changes 128 to
96 to avoid causing a pjsip assertion when making
a call to an endpoint configured with allow=all.
ASTERISK-24367 #close
Review: https://reviewboard.asterisk.org/r/4286/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This updates the documentation for the 'method' configuration option to
be more verbose about the behaviors of values 'unspecified' and
'default'. They do exactly the same thing which is to select the
default as defined by PJSIP which is currently TLSv1.
Review: https://reviewboard.asterisk.org/r/4264/
........
Merged revisions 430145 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Updated the queues.conf.sample file to explicitly state which channel queue
variables are propagated to.
ASTERISK-24267
Reported by: Mitch Claborn
........
Merged revisions 430126 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 430127 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
We only need to hold the context_merge_lock once. Locking it twice will make
many other parts of Asterisk very sad.
ASTERISK-24641 #close
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
If you remove an endpoint/aor from pjsip.conf then do a core reload,
qualifies will continue even though the object are gone. This happens
because nothing clears out the qualify tasks.
This patch unschedules all existing qualify tasks before scheduling
new ones on reload.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4290/
........
Merged revisions 430064 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This patch adds a trailing slash to the category for this test.
No more warning.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4295/
........
Merged revisions 430059 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
After the initial DTMF atxfer call attempt to the transfer target fails to
answer during a blonde transfer, the recall callback channels do not get
setup with information from the initial transferrer channel. As a result,
the recall callback to the transferrer does not have callid, channel
variables, datastores, accountcode, peeraccount, COLP, and CLID setup. A
similar situation happens with the recall callback to the transfer target
but it is less visible. The recall callback to the transfer target does
not have callid, channel variables, datastores, accountcode, peeraccount,
and COLP setup.
* Added missing information to the recall callback channels before
initiating the call. callid, channel variables, datastores, accountcode,
peeraccount, COLP, and CLID
* Set callid of the transferrer channel on the DTMF atxfer controller
thread attended_transfer_monitor_thread().
* Added missing channel unlocks and props unref to off nominal paths in
attended_transfer_properties_alloc().
ASTERISK-23841 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/4259/
........
Merged revisions 430034 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The QUEUESTART log entry has historically acted like a fully booted event
for the queue_log file. When the QUEUESTART entry was posted to the log
was broken by the change made by ASTERISK-15863.
* Made post the QUEUESTART queue_log entry when Asterisk fully boots.
This restores the intent of that log entry and happens after realtime has
had a chance to load.
AST-1444 #close
Reported by: Denis Martinez
Review: https://reviewboard.asterisk.org/r/4282/
........
Merged revisions 430009 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 430010 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Given the following scenario:
* Three SIP phones (A, B, C), all communicating via a proxy with Asterisk
* A call is established between A and B. B performs a SIP attended transfer of
A to C. B sets the call on hold (A is hearing MOH) and dials the extension of
C. While phone C is ringing, B transfers the call (that is, what we typically
call a 'blond transfer').
* When the transfer completes, A hears the ringing of phone C, while B is idle.
In the SIP messaging for the above scenario, a REFER request is sent to
transfer the call. When "sendrpid=yes" is set in sip.conf, Asterisk may send an
UPDATE request to phone C to update party information. This update is sent
directly to phone C, not through the intervening proxy. This has the unfortunate
side effect of providing route information, which is then set on the sip_pvt
structure for C. If someone (e.g. B) is trying to get the call back (through a
directed pickup), Asterisk will send a CANCEL request to C. However, since we
have now updated the route set, the CANCEL request will be sent directly to C
and not through the proxy. The phone ignores this CANCEL according to RFC3261
(Section 9.1).
This patch updates reqprep such that the route is not updated if an UPDATE
request is being sent while the INVITE state is INV_PROCEEDING or
INV_EARLY_MEDIA. This ensures that a subsequent CANCEL request is still sent
to the correct location.
Review: https://reviewboard.asterisk.org/r/4279
ASTERISK-24628 #close
Reported by: Karsten Wemheuer
patches:
issue.patch uploaded by Karsten Wemheuer (License 5930)
........
Merged revisions 429982 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 429983 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429984 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This patch adds the ability for channel drivers to supply presence information
in a similar manner to device state. The patch does not provide any channel
driver implementations, but it does provide the core infrastructure necessary
for channel drivers to provide such information.
The core handles multiple providers of presence state information. Ordering
of presence state is as follows:
INVALID < NOT_SET < AVAILABLE < UNAVAILABLE < CHAT < AWAY < XA < DND
Each provider can trump the previous if it provides a presence state that
supercedes a previous one.
Review: https://reviewboard.asterisk.org/r/4050
ASTERISK-24363 #close
Reported by: Gareth Palmer
patches:
chan_presencestate-428146.patch uploaded by Gareth Palmer (License 5169)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Summaries can't contain XML nodes, as they are defined to contain only text
data.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This patch adds the ability to pass options and a command to MixMontor when
recording a conference using ConfBridge.
New options are -
* record_options: Options to MixMontor, eg: m(), W() etc.
* record_command: The command to execute when recording is over.
* record_file_timestamp: Append the start time to the file name.
These options can also be used with the CONFBRIDGE function, e.g.,
Set(CONFBRIDGE(bridge,record_command)=/path/to/command ^{MIXMONITOR_FILENAME}))
Review: https://reviewboard.asterisk.org/r/4023
ASTERISK-24351 #close
Reported by: Gareth Palmer
patches:
record_command-428838.patch uploaded by Gareth Palmer (License 5169)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Reloading wasn't working correctly because on a reload, the sorcery apply
handler was never being called for unchanged users. So, instead of using
an apply handler, I'm now iterating over all users. Works much more reliably.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4288/
........
Merged revisions 429914 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
startup.
The named ACL code incorrectly destroyed the config options information if loading
of the configuration file failed at startup. This would result in reloading
also failing even if a valid configuration file was put in place.
ASTERISK-23733 #close
Reported by: Richard Kenner
........
Merged revisions 429893 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 429894 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429895 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This won't fix the reported issue but it is an incorrect use of sizeof.
ASTERISK-24566
Reported by: Badalian Vyacheslav
........
Merged revisions 429867 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 429868 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
When the configuration section scheme of chan_dahdi.conf is used (keyword
dahdichan instead of channel) all setvar= options are completely ignored.
No variable defined this way appears in the created DAHDI channels.
* Move the clearing of setvar values to after the deferred processing of
dahdichan.
AST-1378 #close
Reported by: Guenther Kelleter
Patch by: Guenther Kelleter
........
Merged revisions 429825 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 429829 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
After a blond transfer (start attended and hang up)
to a destination that also hangs up without answer,
the Local;1 channel was leaked and would show up on
core show channels. This was happening because the
attended state blond_nonfinal_enter() resetting the
props->transfer_target to null while releasing it's
own reference, which would later prevent props from
releasing another reference during destruction. The
change made here is simply to not assign the target
to NULL.
ASTERISK-24513 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4262/
........
Merged revisions 429826 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 429827 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
ASTERISK-24337 #close
Reported by: Rusty Newton
........
Merged revisions 429804 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 429805 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429806 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
mode.
For the featdmf signaling mode the incoming MF Caller-ID information is
formatted as follows: *${CALLERID(ani2)}${CALLERID(ani)}#*${EXTEN}#
Rather than discarding the ani2 digits, populate the CALLERID(ani2) value
with what is received instead.
AST-1368 #close
Reported by: Denis Martinez
Patches:
extract_ani2_for_featdmf_v11.patch (license #5621) patch uploaded by Richard Mudgett
........
Merged revisions 429783 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 429784 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|