Age | Commit message (Collapse) | Author |
|
This is to support the backwards compatible changes made in the next version
of Asterisk.
........
Merged revisions 420805 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 420808 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420811 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Return the correct value instead of always returning 0 when setting
internal status on unreal channels.
Reported by: Richard Mudgett
........
Merged revisions 420802 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 420803 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The patch to catch channels being shoehorned into Stasis() via external
mechanisms also happens to catch Announcer and Recorder channels
because they aren't known to be stasis-controlled channels in the usual
sense. This marks those channels as Stasis()-internal channels and
allows them directly into bridges.
Review: https://reviewboard.asterisk.org/r/3903/
........
Merged revisions 420795 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 420796 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The unit tests require a sorcery.conf file that has been
set up to store resource lists in memory rather than retrieving
from configuration.
With a setup that is not conducive to running the tests, a fault
in sorcery currently causes Asterisk to crash when attempting to
run any of the tests.
To get around the crash, this adds a function that verifies the
current environment and marks the tests as "not run" if the setup
is not correct.
........
Merged revisions 420779 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Running testsuite tests locally produced no errors, but when
run using the continuous integration framework, crashes occurred.
The crashes occurred due to a refcounting error that had been fixed
for a similar situation.
........
Merged revisions 420758 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
These modules were originally specified as being disabled, as they were
introduced midstream in Asterisk 12. That makes it nicer for folks who are
upgrading to a new release in the middle of Asterisk 12. That's not the case
for Asterisk 13: it's a brand new release. There's no reason to have the
modules disabled by default in that case.
........
Merged revisions 420742 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
If the space left in a stringfield is between 0 and
(alignof(ast_string_field_allocation)-1) adding new data would cause
memory corruption, because we would assume enough space (unsigned
underrun).
Thanks Arnd Schmitter for reporting and finding out the cause!
ASTERISK-23508 #close
Reported by: Arnd Schmitter
Tested by: Arnd Schmitter, JoshE
Review: https://reviewboard.asterisk.org/r/3898/
........
Merged revisions 420680 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 420715 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 420716 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 420717 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 420654 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 420655 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 420656 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 420657 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This patch merely reformats and cleans up a bit of the jitterbuffer
documentation for the wiki.
........
Merged revisions 420639 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This patch gives the optional ability to keep queue rules in RealTime. It is
important to note that with this patch:
(a) Queue rules in RealTime are only examined on module load/reload
(b) Queue rules are loaded both from the queuerules.conf file as well as the
RealTime backend
To inform app_queue to examine RealTime for queue rules, a new setting has been
added to queuerules.conf's general section "realtime_rules". RealTime queue
rules will only be used when this setting is set to "yes".
The schema for the database table supports a rule_name, time, min_penalty, and
max_penalty columns. min_penalty and max_penalty can be relative, if a '-' or
'+' literal is provided. Otherwise, the penalties are treated as constants.
For example:
rule_name, time, min_penalty, max_penalty
'default', '10', '20', '30'
'test2', '20', '30', '55'
'test2', '25', '-11', '+1111'
'test2', '400', '112', '333'
'test3', '0', '4564', '46546'
'test_rule', '40', '15', '50'
which would result in :
Rule: default
- After 10 seconds, adjust QUEUE_MAX_PENALTY to 30 and adjust
QUEUE_MIN_PENALTY to 20
Rule: test2
- After 20 seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust
QUEUE_MIN_PENALTY to 30
- After 25 seconds, adjust QUEUE_MAX_PENALTY by 1111 and adjust
QUEUE_MIN_PENALTY by -11
- After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust
QUEUE_MIN_PENALTY to 112
Rule: test3
- After 0 seconds, adjust QUEUE_MAX_PENALTY to 46546 and adjust
QUEUE_MIN_PENALTY to 4564
Rule: test_rule
- After 40 seconds, adjust QUEUE_MAX_PENALTY to 50 and adjust
QUEUE_MIN_PENALTY to 15
If you use RealTime, the queue rules will be always reloaded on a module
reload, even if the underlying file did not change. With the option disabled,
the rules will only be reloaded if the file was modified.
Review: https://reviewboard.asterisk.org/r/3607/
ASTERISK-23823 #close
Reported by: Michael K
patches:
app_queue.c_realtime_trunk.patch uploaded by Michael K (License 6621)
........
Merged revisions 420624 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 420609 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420610 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 420592 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
ASTERISK-24045
Reported by: Jacob Barber
Review: https://reviewboard.asterisk.org/r/3833/
........
Merged revisions 420577 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 420562 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 420538 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 420536 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 420533 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 420534 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This updates the CEL unit tests for the new information contained in
the attended transfer CEL extra field.
........
Merged revisions 420513 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 420514 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420515 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Replace sip_tls_read() and sip_tcp_read() with a single function and
resolve the poll/wait issue with large SDP payloads.
ASTERISK-18345 #close
Reported by: Stephane Chazelas
Patches:
tcptls_pollv4.diff (license #5835) patch uploaded by Elazar Broad
Review: https://reviewboard.asterisk.org/r/3882/
........
Merged revisions 420434 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 420435 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 420436 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This fixes the json object creation format string and key name for the
BridgeBlindTransfer Stasis event allowing it to be published properly.
........
Merged revisions 420414 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This makes Stasis() event generation for transfer messages follow
validation rules. Currently, ast_json_null() is being used in place of
omitting a key entirely which falls afoul of these validation rules.
https://reviewboard.asterisk.org/r/3892/
........
Merged revisions 420408 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
publishing transfer information.
........
Merged revisions 420387 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420388 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This commit adds the ability for a user to configure
a resource list in pjsip.conf. Subscribing to this
list simultaneously subscribes the subscriber to all
resources listed. This has the potential to reduce
the amount of SIP traffic when loads of subscribers
on a system attempt to subscribe to each others' states.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* Fixed the iax.conf bandwidth option. This is the root cause of
ASTERISK-24150.
* Added checks in iax2_request() to ensure that there are actual formats
requested for the new channel to prevent any more fracks from issues like
ASTERISK-24150. This is a consequence of the iax.conf bandwidth option
not working.
* Fixed struct iax2_codec_pref.order member size mismatch issue when
converting to and from the codec preference order list passed over the
wire. In addition the values sent over the wire are now compatible with
previous Asterisk versions.
* Fixed several issues dealing with the struct iax2_codec_pref members.
Off-by-one, array limit errors, and the order/framing members always need
to be updated together.
* Made iax2_request() setup the channel's native format preference order
according to the user's wishes. The new media format strategy needs the
order specified earler.
* Fixed usage of ast_format_compatibility_bitfield2format(). The function
can return NULL if the bitfield was not associated with a function.
* Deleted dead code iax2_codec_pref_getsize() and
iax2_codec_pref_setsize().
* Made iax2_parse_allow_disallow() and iax2_codec_pref_string() call
iax2_codec_pref_to_cap() instead of inlining it.
* Made IAX_CAPABILITY_MEDBANDWIDTH, IAX_CAPABILITY_LOWBANDWIDTH, and
IAX_CAPABILITY_LOWFREE constants again as they were in Asterisk v1.8.
* Renamed prefs to prefs_global so it won't get confused with the local
pref versions.
* Fixed too small buffer in handle_cli_iax2_show_peer().
* Fixed ast_cli() calls in handle_cli_iax2_show_peer() to output complete
lines.
* Changed struct create_addr_info.prefs to be struct iax2_codec_pref as an
optimization so iax2_request() and iax2_call() do less work.
* Fixed a potential deadlock in ast_iax2_new() on an off-nominal path when
the pbx could not get started.
* Made set_config() setup a local prefs list along side the local
capability format bitfield. Once the config is loaded, then the local
copies are put into the global versions.
* Fix unininialized codec_buf in function_iaxpeer().
ASTERISK-24150 #close
Reported by: Scott Griepentrog
Review: https://reviewboard.asterisk.org/r/3890/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This fixes a class of issues where Stasis applications were not made
aware that their channels were being manipulated or replaced by
external entitiessuch as transfers, AMI commands, or dialplan
applications such as Bridge(). Inconsistent information such as
StasisEnd events with unknown channels as a result of masquerades has
also been corrected. To accomplish these fixes, several new fields
were added to blind and attended transfer messages as well as
StasisStart and BridgeAttendedTransfer Stasis events.
ASTERISK-23941 #close
Review: https://reviewboard.asterisk.org/r/3865/
Review: https://reviewboard.asterisk.org/r/3857/
Review: https://reviewboard.asterisk.org/r/3852/
Review: https://reviewboard.asterisk.org/r/3816/
Review: https://reviewboard.asterisk.org/r/3731/
Review: https://reviewboard.asterisk.org/r/3729/
Review: https://reviewboard.asterisk.org/r/3728/
........
Merged revisions 420325 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
state using SIP.
This module uses the inbound and outbound PUBLISH support to exchange device and mailbox
state between Asterisk instances. Each instance is configured to publish to the other and
requires no intermediary server. The functionality provided is similar to the XMPP and
Corosync support.
Review: https://reviewboard.asterisk.org/r/3780/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This module implements the core parts required for doing outbound PUBLISH.
It takes care of configuration, lifetime management, and authentication.
Additional modules implement the specific events that are published.
Review: https://reviewboard.asterisk.org/r/3780/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Hints that are a pattern match are technically stored in the hint container in
the same fashion as concrete implementations of hints. The pattern matching
hints, however, are not "real" in the sense that things can subscribe to them:
rather, they are stored in the hints container so that when a subscription is
made a "real" hint can be generated for the subscription if one does not yet
exist. The extension state core takes care of this correctly by matching
against non-pattern matching extensions prior to pattern matching extensions.
Because of this, however, the ExtensionStateList AMI action was returning
pattern matching hints when executed. These hints are meaningless from the
perspective of AMI clients: their state will never change, they cannot be
subscribed to, and events would never normally be generated from them. As such,
we now filter these out of the response.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
AMI action responses can (and will) reference AMI events that they return.
These event references and definitions should not be combined with AMI events
raised elsewhere in the code, as they are specifically tied to the AMI action
that raised them.
ASTERISK-24156 #close
Reported by: Rusty Newton
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Change comment.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
When run in offline mode, this would attempt to check the database for
the presence of a type it was going to try to create. I now check the
context to see if we're running in offline mode and change a parameter
accordingly.
........
Merged revisions 407567 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 411514 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* Increased the sippeers useragent max string size to 255.
* Changed the queue_members uniqueid to an auto incremented integer
instead of a string.
* Increased the voicemail_messages BLOB size to LONGBLOB on mysql.
* Fixed the add_tables_for_pjsip config change version downgrade actions
to drop a table it created.
* Adjusted the sample alembic.ini files cdr.ini.sample, config.ini.sample,
and voicemail.ini.sample to give a mysql and postgres sqlalchemy.url
lines.
ASTERISK-23847 #close
Reported by: Stephen More
ASTERISK-23825 #close
Reported by: Stephen More
ASTERISK-23909 #close
Reported by: Stephen More
Review: https://reviewboard.asterisk.org/r/3870/
........
Merged revisions 420211 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
ASTERISK-23818 (lua contexts being overwritten by contexts of the same name in
pbx_config) surfaced because pbx_lua, having the AST_MODFLAG_GLOBAL_SYMBOLS
set, was always force loaded before pbx_config. Since I couldn't find any
reason for pbx_lua to export it's symbols to the rest of Asterisk, I simply
changed the flag to AST_MODFLAG_DEFAULT. Problem solved. What I didn't
realize was that the symbols need to be exported not because Asterisk needs
them but because any external Lua modules like luasql.mysql need the base
Lua language APIs exported (ASTERISK-17279).
Back to ASTERISK-23818... It looks like there's an issue in pbx.c where
context_merge was only merging includes, switches and ignore patterns if
the context was already existing AND has extensions, or if the context was
brand new. If pbx_lua is loaded before pbx_config, the context will exist
BUT pbx_lua, being implemented as a switch, will never place extensions in
it, just the switch statement. The result is that when pbx_config loads,
it never merges the switch statement created by pbx_lua into the final
context.
This patch sets pbx_lua's modflag back to AST_MODFLAG_GLOBAL_SYMBOLS and adds
an "else if" in context_merge that catches the case where an existing context
has includes, switchs or ingore patterns but no actual extensions.
ASTERISK-23818 #close
Reported by: Dennis Guse
Reported by: Timo Teräs
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3891/
........
Merged revisions 420146 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 420147 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 420148 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420149 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
(belongs with r419970)
ASTERISK-24040 #close
Patches:
func_channel.c.diff uploaded by dtryba
Review: https://reviewboard.asterisk.org/r/3781/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This introduces stasis.conf and a mechanism to prevent certain message
types from being published. Internally, this works by preventing the
chosen message types from being created which ensures that those
message types can never be published. This patch also adjusts message
publishers such that message payloads are not created if the related
message type is not available.
ASTERISK-23943 #close
Review: https://reviewboard.asterisk.org/r/3823/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 420099 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420100 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
r420089 | mjordan | 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) | 72 lines
ARI: Add channel technology agnostic out of call text messaging
This patch adds the ability to send and receive text messages from various
technology stacks in Asterisk through ARI. This includes chan_sip (sip),
res_pjsip_messaging (pjsip), and res_xmpp (xmpp). Messages are sent using the
endpoints resource, and can be sent directly through that resource, or to a
particular endpoint.
For example, the following would send the message "Hello there" to PJSIP
endpoint alice with a display URI of sip:asterisk@mycooldomain.org:
ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There
This is equivalent to the following as well:
ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There
Both forms are available for message technologies that allow for arbitrary
destinations, such as chan_sip.
Inbound messages can now be received over ARI as well. An ARI application that
subscribes to endpoints will receive messages from those endpoints:
{
"type": "TextMessageReceived",
"timestamp": "2014-07-12T22:53:13.494-0500",
"endpoint": {
"technology": "PJSIP",
"resource": "alice",
"state": "online",
"channel_ids": []
},
"message": {
"from": "\"alice\" <sip:alice@127.0.0.1>",
"to": "pjsip:asterisk@127.0.0.1",
"body": "Watson, come here.",
"variables": []
},
"application": "testsuite"
}
The above was made possible due to some rather major changes in the message
core. This includes (but is not limited to):
- Users of the message API can now register message handlers. A handler has
two callbacks: one to determine if the handler has a destination for the
message, and another to handle it.
- All dialplan functionality of handling a message was moved into a message
handler provided by the message API.
- Messages can now have the technology/endpoint associated with them.
Various other properties are also now more easily accessible.
- A number of ao2 containers that weren't really needed were replaced with
vectors. Iteration over ao2_containers is expensive and pointless when
the lifetime of things is well defined and the number of things is very
small.
res_stasis now has a new file that makes up its structure, messaging. The
messaging functionality implements a message handler, and passes received
messages that match an interested endpoint over to the app for processing.
Note that inadvertently while testing this, I reproduced ASTERISK-23969.
res_pjsip_messaging was incorrectly parsing out the 'to' field, such that
arbitrary SIP URIs mangled the endpoint lookup. This patch includes the
fix for that as well.
Review: https://reviewboard.asterisk.org/r/3726
ASTERISK-23692 #close
Reported by: Matt Jordan
ASTERISK-23969 #close
Reported by: Andrew Nagy
........
r420090 | mjordan | 2014-08-05 15:16:37 -0500 (Tue, 05 Aug 2014) | 2 lines
Remove automerge properties :-(
........
r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue, 05 Aug 2014) | 2 lines
test_message: Fix strict-aliasing compilation issue
........
Merged revisions 420089-420090,420097 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
format.c: Add reason comments for the format_list ordering.
........
Merged revisions 420054 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Or any combination of codecs that includes Opus.
ASTERISK-24107 #close
Review: https://reviewboard.asterisk.org/r/3885/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This adds the ability to call CHANNEL(recvport) on chan_sip
channels to see the port on which an INVITE was received.
ASTERISK-24040 #close
Reported by dtryba
Patches:
dialplan_functions.patch uploaded by dtryba (License #6628)
Review: https://reviewboard.asterisk.org/r/3781
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The documentation for these commands did not make it clear that they could
accept expressions and functions. Modified to make this clear, but tried
not to be overly explicit.
ASTERISK-21178 #close
Reported by: Rusty Newton
Tested by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/3854
........
Merged revisions 419942 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 419943 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 419944 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This adds a large swath of response documentation for PJSIPShowEndpoint
and PJSIPShowEndpoints AMI commands. It relies heavily on the existing
text in the configInfo documentation via xi:include tags to avoid
documentation duplication.
Review: https://reviewboard.asterisk.org/r/3888/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Now when running PJSIPShowEndpoint, you will receive a
ContactStatusDetail for each bound contact that Asterisk
is qualifying. This information includes the URI of the
contact, current reachability, and roundtrip time.
AFS-91 #close
Reported by Mark Michelson
Review: https://reviewboard.asterisk.org/r/3797
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|