Made agent able to interrupt the alerting beep playback with DTMF. Any
digit can interrupt if the call does not need to be acknowledged. Only
the first digit of the acknowledgement can interrupt if the call needs to
be acknowledged. The agent interrupting the alerting playback builds on
the ASTERISK-24447 patch because it knows what digit interrupted the
playback and needs to be able to pass that digit to the DTMF hook digit
collection code.
ASTERISK-24257 #close
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4123/
........
Merged revisions 427508 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When handling incoming messages we determine if it is associated with
a dialog. If so we use that to determine what serializer and endpoint
to use for the message. Previously this would pass the endpoint to the
endpoint lookup module to actually place the endpoint completely on the
message. For in-dialog responses, however, this did not occur as
dialog processing took over and the endpoint lookup did not occur.
This change just places the endpoint in the expected spot immediately
instead of relying on the endpoint lookup module. In-dialog responses
thus have the expected endpoint.
AST-1459 #close
Review: https://reviewboard.asterisk.org/r/4146/
........
Merged revisions 427490 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427491 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Testing has shown repeatedly that PJSIP's default behavior of switching
automatically to TCP for large messages can cause issues. The most common
issues are that devices that we are communicating with do not handle the
switch to TCP gracefully, thus causing situations such as broken calls or
broken subscriptions. Now, in order to have this behavior happen, you must
opt into it. The sample file has been updated to warn that enabling the
TCP switch behavior may cause issues for you, so use at your own risk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a config file is read, an unescaped semicolon signals comments which are
stripped from the value before it's stored. Escaped semicolons are then
unescaped and become part of the value. Both of these behaviors are normal
and expected. When the config is serialized either by 'dialplan save' or
AMI/UpdateConfig however, the now unescaped semicolons are written as-is.
If you actually reload the file just saved, the unescaped semicolons are
now treated as start of comments.
Since true comments are stripped on read, any semicolons in
ast_variable.value must have been escaped originally. This patch
re-escapes semicolons in ast_variable.values before they're written to
file either by 'dialplan save' or config/ast_config_text_file_save which
is called by AMI/UpdateConfig. I also fixed a few pre-existing formatting
issues nearby in pbx_config.c
Tested-by: George Joseph
ASTERISK-20127 #close
Review: https://reviewboard.asterisk.org/r/4132/
........
Merged revisions 427275 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When merging from 12 to 13 there were conflicts,
I mistakenly had the loop run ast_closestream(others[0])
when it should be ast_closestream(others[x]).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a packet exceeds the MTU, pjproject will switch from UDP to TCP. In
some circumstances (on some networks), this can cause some issues with
messages not getting sent to the correct destination - and can also cause
connections to get dropped due to quirks in pjproject deciding to
terminate TCP connections with no messages.
While fixing the routing/messaging issues is important, having a
configuration option in Asterisk that tells pjproject to not switch over
to TCP would be useful. That way, if some glitch is discovered on some
other network/site, we can at least disable the behavior until a fix is
put into place.
AFS-197 #close
Review: https://reviewboard.asterisk.org/r/4137/
........
Merged revisions 427129 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the res_stasis module is unloaded, it will dispose of the apps_registry
container. This is a problem if an ARI operation is in flight that attempts
to use the registry, as the shutdown occurs in a separate thread. This patch
adds some sanity checks to the various routines that access the registry which
cause the operations to fail if the apps_registry does not exist.
Crash caught by the Asterisk Test Suite.
........
Merged revisions 426995 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A question arose as to whether a .pem file
could be provided in place of the .crt and
.key files in a PJSIP TLS configuration. I
tested this and discovered that although a
cert will be read from the pem file, a key
will not, and thus the priv_key_file entry
is still required. This update to the fine
documentation clarifies the option usage.
AST-1448 #close
Review: https://reviewboard.asterisk.org/r/4129/
Reported by: John Bigelow
........
Merged revisions 426928 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This updates the status of the outbound registration
to reflect when it has been unregistered. Since the
registration is unregistered but is not stopped, the
registration schedule remains active as before. The
patch also updates the documentation of both the AMI
and CLI commands.
ASTERISK-24411 #close
Review: https://reviewboard.asterisk.org/r/4119/
Reported by: John Bigelow
patches:
unregister-patch1.txt uploaded by John Bigelow (License 5091)
........
Merged revisions 426923 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When r426594 was made, it did not take into account a unit test that verified
that the function properly populated the unsupported buffer. The function
would previously memset the buffer if it detected it had any contents; since
this function can now be called iteratively on successive headers, the unit
tests would now fail. This patch updates the unit tests to reset the buffer
themselves between successive calls, and updates the documentation of the
function to note that this is now required.
........
Merged revisions 426858 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 426860 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 426863 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426865 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, it is possible for some subscriptions to get into a NULL state. When
this occurs and the PJSIPShowSubscriptionsInbound ami action is issued and a
device is subscribed for extension state then the associated subscription state
object can't be located. The code then attempts to dereference a NULL object.
Added a NULL check to avoid the problem.
Reported by: John Bigelow
........
Merged revisions 426779 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When removing the qualify_frequency from an AoR or a contact the statistics
shown when issuing "pjsip show aors" from the CLI are incorrect. This patch
deletes the contact's status object from sorcery, disassociating it from the
contact, if the qualify_freqency is removed from configuration.
ASTERISK-24462 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4116/
........
Merged revisions 426755 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The feature_automonitor() and feature_automixmonitor() functions were not
locking the channel around ast_get_chan_features_general_config().
Accessing the channel datastore list without the channel locked is a good
way to corrupt the list or follow the pointer chain into oblivion.
........
Merged revisions 426531 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a channel is imparted to a bridge, the invocation of the function may
provide an ast_bridge_features struct. Upon passing this to ast_bridge_impart,
the caller must assume that ownership has passed to the function, as in all
paths the function destroys the struct prior to returning (as its purpose is
to configure the behavior of the channel while in the bridge). On one off
nominal path - where the channel already has a PBX thread - the struct was not
being destroyed.
This patch fixes that glitch.
ASTERISK-24437 #close
Reported by: Scott Griepentrog
........
Merged revisions 426431 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Because opus transcoding support cannot be included in the standard Asterisk
distribution, a few codec_opus implementations have popped up. To make it
easier for people to drop in opus support in their own installations, this
patch adds configure checks for libopus.
Review: https://reviewboard.asterisk.org/r/4106/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In res_phoneprov, unloading the module first destroys the http_routes
container, followed by the users. However, users may have a route in
the http_routes container; the validity of this container is not checked
in the users destructor. Hence, we hit an assert as the container has already
been set to NULL.
This patch does two things:
(1) It adds a sanity check in the user destructor (because why not)
(2) It switches the order of destruction, so that users are disposed of prior
to the HTTP routes they may hold a reference to.
Note that this crash was caught by the Test Suite (go go testing!)
........
Merged revisions 426174 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes a Segmentation fault introduced in r419044 "media formats: re-architect
handling of media for performance improvements".
The problem is that codec_dahdi was using core_src_codec and core_dst_codec in the
ast_translator structure when these fields were never set. Now instead of trying to map
the new core codec descriptions to the way DAHDI defines different codecs, we will store
the DAHDI specific formats in 'struct translator' directly so we can refer to them without
mapping.
This also allows us to remove the "global_format_map" structure, since we can now query
the list of translators directly to make sure we do not ever register a DAHDI based
translator for a specific path more than once and eliminate the need to keep the list and
the map in sync.
ASTERISK-24435 #close
Reported by: Marian Koniuszko
Review: https://reviewboard.asterisk.org/r/4105/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fix the AMI Status action read and write translation path strings from
growing for each channel in the status event list by reseting the ast
string given to ast_translate_path_to_str() to fill in the given
translation path.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
