Added a new PJSIP global setting called norefersub.
Default is true to keep support working as before.
res_pjsip_refer: Configures PJSIP norefersub capability accordingly.
Checks the PJSIP global setting value.
If it is true (default) it adds the norefersub capability to PJSIP.
If it is false (disabled) it does not add the norefersub capability
to PJSIP.
This is useful for Cisco switches that do not follow RFC4488.
ASTERISK-28375 #close
Reported-by: Dan Cropp
Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9
One of the downaides of having things like test configuration
in the git repo is that it can't be changed at runtime. You have
to create a review for the changes and merge it mefore it will
take effect.
This review moves the data currently held in
tests/CI/periodic-dailyTestGroups.json and
tests/CI/gateTestGroups.json into a Jenkins Config File attached
to the job definitions. This allows us to alter it from the
Jenkins UI at runtime. The original files stay in the repo
as documentation.
Change-Id: I14b9702f6285ce1fb2420287ba0e7d3b59109763
Because the per-mailbox options are the last thing on a line, don't look
for or stomp on any subsequent commas.
ASTERISK-27935 #close
Reported by: Sébastien Duthil
Change-Id: I07b2eb4a33c303d0c7114d5b906f8c067c60a153
Because hyphens are not matched literally in Asterisk dialplan, we need
to ignore them in our candidate extensions as well.
ASTERISK-17695 #close
Reported by: test011
Change-Id: I227f02301577b1633e8a55b9fe9dc149935c03f0
If Asterisk crashes while a VM directory is locked, lock files in the VM
spool directory will not get properly cleaned up. We now clear them on
module load.
ASTERISK-20207 #close
Reported by: Steven Wheeler
Change-Id: If40ccd508e2f6e5ade94dde2f0bcef99056d0aaf
Reset the internal counter that the AEL2 compiler uses for unique label
names before compiling. This keeps dialplan labels consistent across
reloads assuming the AEL2 has not changed.
ASTERISK-17799 #close
Reported by: Kirill Katsnelson
Change-Id: I30b3cc887d1ee0644d3f341e2fef16f525d7fae5
In AEL2, if a 'for' statement contains macro* calls, like:
for (&iterator(${TRY},A); "${A}" != ""; &iterate(A)) {
The AEL2 parser will translate these into calls to the deprecated Macro
dialplan application and use the antiquated pipe delimiter.
Instead, convert these into calls to the Gosub dialplan application and
use commas as argument separators.
ASTERISK-18593 #close
Reported by: Luke-Jr
* 'macro' in this context means AEL2 macros, not the 'Macro' application
Change-Id: I3d73716033b8e3e42e0209d355bf5f10c97045fc
When generating the regular expression that matches against existing
extensions, we need to escape literal characters that can also be
regular expression metacharacters. This was already being done for '*'
but we need to do the same for '+'.
In passing, remove some unreachable code - strcmp() is already run
immediately when entering extension_matches().
ASTERISK-14939 #close
Reported by: klaus3000
Change-Id: I8d2cccb3479168fba1b0a6704c52198b396468f1
If the target of a Goto is a label that starts with a number, we
erroneously treat the leading digits as a priority.
ASTERISK-20182 #close
Reported by: Janu
Change-Id: Ia78408c0805a729103917247ecfc802f6fafc94b
This changes the way that we handle adding changes to CHANGES and
UPGRADE.txt. The reason for this is because whenever someone needed to
make a change to one of these files and someone else had already done
so, you would run into merge conflicts. With this new setup, there will
never be merge conflicts since all changes will be documented in the
doc/<file>-staging directory. The release script is now responsible for
merging all of these changes into the appropriate files.
There is a special format that these files have to follow in order to be
parsed. The files do not need to have a meaningful name, but it is
strongly recommended. For example, if you made a change to pjsip, you
may have something like this "res_pjsip_relative_title", where
"relative_title" is something more descriptive than that. Inside each
file, you will need a subject line for your change, followed by a
description. There can be multiple subject lines. The file may look
something like this:
Subject: res_pjsip
Subject: Core
A description that explains the changes made and why. The release
script will handle the bulleting and section separators!
You can still separate with new lines within your description.
The headers ("Subject" and "Master-Only") are case sensative, but the
value for "Master-Only" ("true" or "True") is not.
For more information, check out the wiki page:
https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt
ASTERISK-28111 #close
Change-Id: I19cf4b569321c88155a65e9b0b80f6d58075dd47
When extconfig.conf file is parsed, the code previously searched for
character comma without verifying if error (null or blank). This caused
a segmentation error.
Change-Id: Id76b452d8f330d11c2742c37232761ad71472a8b
When the dtmf_mode on an endpoint is configured as "auto_info"
Asterisk will produce an inband DTMF tone alongside an INFO
message when sending DTMF.
ASTERISK-28371
Change-Id: I1380b82f006e110a1b83fbb50c9873edd13a5d9a
It was difficult to check the channel's current application and
parameters using ARI for current channels. Added app_name, app_data
items to show the current application information.
ASTERISK-28343
Change-Id: Ia48972b3850e5099deab0faeaaf51223a1f2f38c
The new option disables dev mode, TEST_FRAMEWORK and
MALLOC_DEBUG making the build more production-like.
Change-Id: Ieb72497d4d91d5416684aaed702cc3f532099738
The compiler complained about a couple of variables that weren't
initialized but were being used. Initializing them to NULL resolves the
warnings/errors.
ASTERISK-28362 #close
Change-Id: I6243afc5459b416edff6bbf571b0489f6b852e4b
It looks like we're not properly calculating jitter values on received
video streams. This patch enables the code that does jitter calculations
for those streams.
Change-Id: Iaac985808829c8f034db8c57318789c4c8c11392
When trying to upgrade using alembic, a couple different errors kept
popping up that prevented the upgrade. An additional parameter was
needed when changing the schema for mwi_subscribe_replaces_unsolicited
from an integer to an enum. When changing from a string to an enum, the
type needed to be cast for postgresql. The other issue was a parameter
being used during column creation that did not exist.
After fixing the upgrade process, it revealed errors with the downgrade
process. One was a variable not being defined in the downgrade function,
and the other was tables not existing when using MySQL. This was due to
a context check that should have encompassed MySQL, but in the end was
not doing so.
Change-Id: Ib4d70cf3ce5080023a50be496272a777b55d6c8e
Added ARI resource for channel statistics.
GET /ari/channels/{channelId}/rtp_statistics : It returns given
channel's rtp statistics detail.
ASTERISK-28320
Change-Id: I4343eec070438cec13f2a4f22e7fd9e574381376
This is the first step in changing the release process so that changes
made to the CHANGES and UPGRADE.txt files do not result in merge
conflicts every time multiple people modify these files. The changes
made will go in these new directories: doc/CHANGES-staging and
doc/UPGRADE-staging. The README.md files explain how things will work,
but here's a little overview. When you make a change that would go in
either CHANGES or UPGRADE.txt, this should instead be documented in a
new file in the doc/CHANGES-staging or doc/UPGRADE-staging directory,
respectively. The format will look like this:
Subject: res_pjsip
A description that explains the changes made and why. The release
script will handle the bulleting and section separators! The
'Subject:' header is case-sensitive.
You can still separate with new lines within your description.
Subject: res_ari
Master-Only: true
You can have more than one subject, and they don't have to be the
same! Also, the 'Master-Only' header should always be true and is
also case-sensitive (but the value is not - you can have 'true' or
'True'). This header will only ever be present in the master branch.
For more information, check out the wiki page:
https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt
This is an initial change for ASTERISK_28111. Functionally, this will
make no difference, but it will prep the directories for when the
changes from CHANGES and UPGRADE.txt are extracted.
Change-Id: I8d852f284f66ac456b26dcb899ee46babf7d15b6
The next usage of PJSIP_PARSE_URI will crash asterisk
${PJSIP_PARSE_URI(tel:+1234567890,host)}
or
${PJSIP_PARSE_URI(192.168.1.1:5060,host)}
The function pjsip_parse_uri successfully parses then, but returns
struct pjsip_other_uri *.
This patch restricts parsing only SIP/SIPS URIs.
Change-Id: I16f255c2b86a80a67e9f9604b94b129a381dd25e
It was a copy/paste of the QUEUE_MEMBER_COUNT function's synopsis.
ASTERISK-20986 #close
Reported by: Olivier Krief
Change-Id: If51ec481feb35824a4e78ab5600b197b819b10be
When notifying a manager session that new events were available
the same lock was used that was also held when doing things within
the session (such as sending events out). If the manager session
blocked for a period of time this would cause a back up of messages
in Stasis and would also block any other sessions from receiving
events.
This change adds a separate lock to the manager session which is
strictly used for notifying it that new events are available.
ASTERISK-28350
Change-Id: Ifbcac007faca9ad0231640f5e82a6ca9228f261b
