Occasionally, the manager module would get an "INTERNAL_OBJ: bad magic
number" error on a "core restart gracefully" command if an AMI connection
is established.
* Added ao2_global_obj protection to the sessions global container.
* Fixed the order of unreferencing a session object in session_destroy().
* Removed unnecessary container traversals of the white/black filters
during session_destructor().
(closes issue AST-1242)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/3144/
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When support for a realtime sorcery module was added in revision 386731, the
wrong property was accidentally used for setting the column name to be updated
in the database table. This patch fixes the typo.
(closes issue ASTERISK-23177)
Reported by: Denis
Tested by: Denis
Patches:
asterisk-23177-use-field-name.diff by Michael L. Young (license 5026)
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r406294 | mmichelson | 2014-01-23 15:00:24 -0600 (Thu, 23 Jan 2014) | 11 lines
Fix presence body errors found during testing:
* PIDF bodies were reporting an "open" state in many cases where
it should have been reporting "closed"
* XPIDF bodies had XML nodes placed incorrectly within the hierarchy.
* SIP URIs in XPIDF bodies did not go through XML sanitization
* XML sanitization had some errors:
* Right angle bracket was being replaced with "&rt;" instead of ">"
* Double quote, apostrophe, and ampersand were not being escaped.
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r406295 | mmichelson | 2014-01-23 15:09:35 -0600 (Thu, 23 Jan 2014) | 11 lines
Fix presence body errors found during testing:
* PIDF bodies were reporting an "open" state in many cases where
it should have been reporting "closed"
* XPIDF bodies had XML nodes placed incorrectly within the hierarchy.
* SIP URIs in XPIDF bodies did not go through XML sanitization
* XML sanitization had some errors:
* Right angle bracket was being replaced with "&rt;" instead of ">"
* Double quote, apostrophe, and ampersand were not being escaped.
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Several of the playback error messages for invalid media input in
res_stasis_playback.c had the media name and channel name reversed.
They now correctly identify the channel name and media name.
Reported by: skrusty
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Making the help text for both more explicit regarding the format of mailbox identifiers. i.e. clarifying the format for app_voicemail mailboxes vs mailboxes from external MWI sources through modules such as res_external_mwi.
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The PJSIP header parsing function (pjsip_parse_hdr) can generate more
than one header instance from a single header field. These header
instances exist as a list attached to the returned header and must be
handled appropriately when they are added to a message or else only the
first header instance will be used. This changes the linked list
functions used in outbound proxy code to merge the lists properly.
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This adds back in support for specifying channel variables during an
originate without compromising the ability to specify query parameters
in the JSON body. This was accomplished by generating the body-parsing
code in a separate function instead of being integrated with the URI
query parameter parsing code such that it could be called by paths with
body parameters. This is transparent to the user of the API and
prevents manual duplication of code or data structures.
(closes issue ASTERISK-23051)
Review: https://reviewboard.asterisk.org/r/3122/
Reported by: Matt Jordan
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This change adds improvements to support for allow=all in
pjsip.conf so that it functions as intended. Previously,
the allow/disallow socery configuration would set & clear
codecs from the media.codecs and media.prefs list, but if
all was specified the prefs list was not updated. Then a
call would fail when create_outgoing_sdp_stream() created
an SDP with no audio codecs.
A new function ast_codec_pref_append_all() is provided to
add all codecs to the prefs list - only those not already
on the list. This enables the configuration to specify a
codec preference, but still add all codecs, and even then
remove some codecs, as shown in this example:
allow = ulaw, alaw, all, !g729, !g723
Also, the display order of allow in cli output is updated
to match the configuration by using prefs instead of caps
when generating a human readable string.
Finally, a change to create_outgoing_sdp_stream() skips a
codec when it does not have a payload code instead of the
call failing.
(closes issue ASTERISK-23018)
Reported by: xrobau
Review: https://reviewboard.asterisk.org/r/3131/
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Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue.
Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample.
(issue ASTERISK-23061)
(issue ASTERISK-23028)
(issue ASTERISK-23046)
(issue ASTERISK-23027)
(closes issue ASTERISK-23061)
(closes issue ASTERISK-23028)
(closes issue ASTERISK-23046)
(closes issue ASTERISK-23027)
Reported by: Eugene, Jeremy Laine, Denis Pantsyrev
Patches:
transferred.patch uploaded by Jeremy Laine (license 6561)
hyphen.patch uploaded by Jeremy Laine (license 6561)
sip.conf.sample.patch uploaded by Eugene (license 6360)
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If an endpoint had previously dynamically registered a contact and the contact
information was successfully stored in astdb then upon restart the qualify
notifications would not be sent out if the qualify_frequency was set. This was
due to the fact that only permanent contacts were being checked and scheduled
for qualifies on startup. Modified the code to check and schedule all
registered contacts at startup.
(closes issue ASTERISK-23062)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/3124/
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When path support was added and contacts were made available during
request creation and transmission, the code path used by outbound
qualify support was not modified correctly and was causing request
creation to fail. This ensures that outbound request creation with only
a contact and no dialog, endpoint, or uri can succeed which restores
qualify support.
Reported by: gtjoseph
Reported by: kharwell
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This adds Path support to chan_pjsip in res_pjsip_path.c with minimal
additions in res_pjsip_registrar.c to store the path and additions in
res_pjsip_outbound_registration.c to enable advertisement of path
support to registrars and intervening proxies.
Path information is stored on contacts and is enabled via Address of
Record (AoRs) and Registration configuration sections.
While adding path support, it became necessary to be able to add SIP
supplements that handled messages outside of sessions, so a framework
for handling these types of hooks was added in parallel to the
already-existing session supplements and several senders of
out-of-dialog requests were refactored as a result.
(closes issue ASTERISK-21084)
Review: https://reviewboard.asterisk.org/r/3050/
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Adds the following AMI commands:
PUT mailboxes/mailboxName
modifies mailbox state and implicitly creates new mailboxes
GET mailboxes/mailboxName
retrieves a JSON representation of a single mailbox if it exists
GET mailboxes
retrieves a JSON array of all mailboxes
DELETE mailbox/mailboxName
deletes a mailbox
Note that res_mwi_external must be loaded for these functions to
actually do anything.
Review: https://reviewboard.asterisk.org/r/3117/
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The per console verbose level feature as previously implemented caused a
large performance penalty. The fix required some minor incompatibilities
if the new rasterisk is used to connect to an earlier version. If the new
rasterisk connects to an older Asterisk version then the root console
verbose level is always affected by the "core set verbose" command of the
remote console even though it may appear to only affect the current
console. If an older version of rasterisk connects to the new version
then the "core set verbose" command will have no effect.
* Fixed the verbose performance by not generating a verbose message if
nothing is going to use it and then filtered any generated verbose
messages before actually sending them to the remote consoles.
* Split the "core set debug" and "core set verbose" CLI commands to remove
the per module verbose support that cannot work with the per console
verbose level.
* Added a silent option to the "core set verbose" command.
* Fixed "core set debug off" tab completion.
* Made "core show settings" list the current console verbosity in addition
to the root console verbosity.
* Changed the default verbose level of the 'verbose' setting in the
logger.conf [logfiles] section. The default is now to once again follow
the current root console level. As a result, using the AMI Command action
with "core set verbose" could again set the root console verbose level and
affect the verbose level logged.
(closes issue AST-1252)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/3114/
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We were not including an authentication challenge when sending a 401 response
to unmatched endpoints. This was due to the conversion to use a vector for
authentication section names on an endpoint. The vector for artificial endpoints
was empty, resulting in the challenge being sent back containing no challenges.
This is worked around by placing a bogus value in the artificial endpoint's auth
vector. This value is never looked up by anything, since they instead will directly
call ast_sip_get_artificial_auth().
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This change fixes a few memory leaks that were found based
on a mailing list post.
1. Some JSON response messages were never freed. This was
caused by the documentation stating that message references
were stolen when in reality they were not. The code now follows
the documentation and usage has been updated.
2. HTTP response headers were never freed.
3. The variable list for wildcards paths was never freed.
(closes issue ASTERISK-23128)
Reported by: Kenneth Watson (on list)
Review: https://reviewboard.asterisk.org/r/3119/
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In https://reviewboard.asterisk.org/r/3057/, applications and functions that
manipulate CDRs were made to interact over Stasis. This was done to
synchronize manipulations of CDRs from the dialplan with the updates the
engine itself receives over the message bus.
This change rested on a faulty premise: that messages published to the CDR
topic or to a topic that forwards to the CDR topic are synchronized with the
messages handled by the CDR topic subscription in the CDR engine. This is not
the case. There is no ordering guaranteed for two messages published to the
same topic; ordering is only guaranteed if a message is published to the same
subscriber.
Stasis was modified in r405311 to allow a publisher to synchronize on the
subscriber. This patch uses that API to synchronize the CDR publishers with
the CDR engine message router, which maintains the overall topic subscription.
(closes issue ASTERISK-22884)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3099/
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This patch adds an API call to Stasis that allows a publisher to publish a
stasis message that will not return until a specific subscriber handles the
message. Since a subscriber can have their own forwarding topic which orders
messages from many topics, this allows a publisher who knows of that subscriber
to synchronize to that subscriber regardless of the forwarding relationships
between topics.
This is of particular use for dialplan applications that need to synchronize
on a particular subscriber's handling of a message.
(issue ASTERISK-22884)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3099/
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Richard Mudgett
Mark Michelson
Michael L. Young
Scott Griepentrog
Kinsey Moore
Rusty Newton
Walter Doekes
Damien Wedhorn
Jonathan Rose
Sean Bright
Kevin Harwell
Joshua Colp
Matthew Jordan