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sample call #97

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sample call #97

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@gtj5700 gtj5700 commented Oct 17, 2023

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gtjoseph and others added 30 commits July 17, 2023 08:46
The app_queue module provides both an AMI action and a CLI command
to change the priority of a caller in a queue. Up to now this change
of priority has only been reflected to new callers into the queue.

This change adds an "immediate" option to both the AMI action and
CLI command which immediately applies the priority change respective
to the other callers already in the queue. This can allow, for example,
a caller to be placed at the head of the queue immediately if their
priority is sufficient.

Resolves: #202

UserNote: The 'queue priority caller' CLI command and
'QueueChangePriorityCaller' AMI action now have an 'immediate'
argument which allows the caller priority change to be reflected
immediately, causing the position of a caller to move within the
queue depending on the priorities of the other callers.
c3ff464 removed the [iaxtel700] context but neglected to remove
references to it.

This commit addresses that and also removes iaxtel and freeworlddialup
references from other config files.
This adds an example to the XML documentation clarifying usage
of the CUT function to address a common misusage.
Fixes #221

UserNote: res_pjsip now allows TLS v1.3 to be enabled if supported by
the underlying PJSIP library. The bundled version of PJSIP supports
TLS v1.3.
In 8d6fdf9 invisible bridges were
skipped but that lead to producing metrics with no name and no help.

Keep track of the number of metrics configured and then only emit these.
Add a basic testcase that verifies that there is no '(NULL)' in the
output.

ASTERISK-30474
sig_analog allows users to flash and use the three-way dial
tone as a primitive hold function, simply by never timing
it out.

Some systems allow this dial tone to time out to silence,
so the user is not annoyed by a persistent dial tone.
This option allows the dial tone to time out normally to
silence.

ASTERISK-30004 #close
Resolves: #205

UserNote: The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.
In the case where mute was called on a channel that had no
audiohooks the code was not unlocking the channel, resulting
in a deadlock.

Resolves: #233
Currently, if an FXS channel is still off hook when
all calls on the line have hung up, the user is provided
reorder tone until going back on hook again.

In addition to not reflecting what most commercial switches
actually do, it's very common for switches to automatically
reoriginate for the user so that dial tone is provided without
the user having to depress and release the hookswitch manually.
This can increase convenience for users.

This behavior is now supported for kewlstart FXS channels.
It's supported only for kewlstart (FXOKS) mainly because the
behavior doesn't make any sense for ground start channels,
and loop start signalling doesn't provide the necessary DAHDI
event that makes this easy to implement. Likely almost everyone
is using FXOKS over FXOLS anyways since FXOLS is pretty useless
these days.

ASTERISK-30357 #close

Resolves: #224

UserNote: The autoreoriginate setting now allows for kewlstart FXS
channels to automatically reoriginate and provide dial tone to the
user again after all calls on the line have cleared. This saves users
from having to manually hang up and pick up the receiver again before
making another call.
This change adds support for refers that are not session based. It
includes a refer implementation for the PJSIP technology which results
in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be
triggered using the new ARI endpoint `/endpoints/refer`.

Resolves: asterisk#71

UserNote: There is a new ARI endpoint `/endpoints/refer` for referring
an endpoint to some URI or endpoint.
In some cases I have yet to determine some stasis messages may
be created without a channel snapshot. This change adds some
tolerance to this scenario, preventing a crash from occurring.
The default is 32 with 8 being used by pjproject itself.  Recent
commits have put us over the limit resulting in assertions in
pjproject.  Since this value is used in invites, dialogs,
transports and subscriptions as well as the global pjproject
endpoint, we don't want to increase it too much.

Resolves: #255
The documentation for PJSIP_HEADERS claims that
prefix is optional, but in the code it is actually not.
However, there is no inherent reason for this, as users
may want to retrieve all header names, not just those
beginning with a certain prefix.

This makes the prefix optional for this function,
simply fetching all header names if not specified.
As a result, the documentation is now correct.

Resolves: #230

UserNote: The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
Added a new boolean configuration flag -
`order_multi_row_results_by_initial_column` - to both res_pgsql.conf
and res_config_odbc.conf that allows the administrator to disable the
explicit `ORDER BY` that was previously being added to all generated
SQL statements that returned multiple rows.

Fixes: #179
If the contact_user is configured on the endpoint it will now be set on the local Contact header URI for incoming calls. The contact_user has already been set on the local Contact header URI for outgoing calls.

Resolves: #226
Fixes dependency solutions in install_prereq for Debian aarch64
platforms. install_prereq was attempting to forcibly install 32-bit
armhf packages due to the aptitude search for dependencies.

Resolves: asterisk#37
This reverts commit 617dad4.

apps/app_stack.c: Revert buggy gosub patch

This seems to break the case when a predial macro calls a gosub.
When the gosub calls return, the Return function outputs:

app_stack.c:423 return_exec: Return without Gosub: stack is empty

This returns -1 to the calling macro, which returns to app_dial
and causes the call to hangup instead of proceeding with the macro
that invoked the gosub.

Resolves: #253
This adds support for Called Subscriber Held for FXS
lines, which allows users to go on hook when receiving
a call and resume the call later from another phone on
the same line, without disconnecting the call. This is
a convenience mechanism that most real PSTN telephone
switches support.

ASTERISK-30372 #close

Resolves: #240

UserNote: Called Subscriber Held is now supported for analog
FXS channels, using the calledsubscriberheld option. This allows
a station  user to go on hook when receiving an incoming call
and resume from another phone on the same line by going on hook,
without disconnecting the call.
* Fixed issue with the script not parsing the new tag format for
  certified releases.  The format changed from certified/18.9-cert5
  to certified-18.9-cert5.

* Fixed issue where the asterisk version wasn't being considered
  when looking for cached versions.

Resolves: #263
Add quoting around the ps_endpoints 100rel column in the ALTER
statements.  Although alembic doesn't complain when generating
sql statements, postgresql does (rightly so).

Resolves: #274
Resolve for loop initial declarations added in cli changes.

Resolves: #275
If the called party hangs up while digits are being
sent, -1 is returned to indicate so, but app_dial
was not checking the return value, resulting in
the hangup being lost and looping forever until
the caller manually hangs up the channel. We now
abort if digit sending fails.

ASTERISK-29428 #close

Resolves: #281
Handle session interval lower than endpoint's configured minimum timer
when sending first answer. Timer setting is checked during this step and
needs to handled appropriately.
Before this change, no response was sent at all. After this change a
response with 422 Session Interval too small is sent to UAC.
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If you don't want it cherry-picked, please add a comment stating "No cherry-picks required" so we don't keep asking.

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sangoma-oss-cla bot commented Feb 22, 2024

CLA assistant check
Thank you for your submission! We really appreciate it. Like many open source projects, we ask that you all sign our Contributor License Agreement before we can accept your contribution.
16 out of 18 committers have signed the CLA.

✅ gtjoseph
✅ jcolp
✅ seanbright
✅ phoneben
✅ zecke
✅ InterLinked1
✅ maximilianfridrich
✅ MikeNaso
✅ creslin287
✅ nphantom
✅ mbradeen
✅ btriller
✅ Tinet-mucw
✅ eduardomazolini
✅ jkroonza
✅ vnovy
❌ jxmx
❌ gtj5700
You have signed the CLA already but the status is still pending? Let us recheck it.

@gtjoseph gtjoseph closed this Mar 5, 2024
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