AMI/ARI are getting a few enhancements in the next release of Asterisk 13. Per
semantic versioning, that warrants a bump in the minor version number, as it
reflects a backwards compatible change. Hence, this commit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Optimistic SRTP is the ability to enable SRTP but not have it be
a fatal requirement. If SRTP can be used it will be, if not it won't be.
This gives you a better chance of using it without having your sessions
fail when it can't be.
Encrypt all the things!
Review: https://reviewboard.asterisk.org/r/3992/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This allows for a configurable number of attempts for a transferer
to dial an extension to transfer the call to. For Asterisk 13, the
default values are such that upgrading between versions will not
cause a behaivour change. For trunk, though, the defaults will be
changed to be more user-friendly.
Review: https://reviewboard.asterisk.org/r/4167
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch addresses a few issues:
1) The order of Dial events have been changed when performing a call forward.
The order has now been altered to
1) Dial begins dialing channel A.
2) When A forwards the call to B, we issue the dial end event to channel
A, indicating the dial is being canceled due to a forward to B.
3) When the call to channel B occurs, we then issue a new dial begin to
channel B.
2) Call forwards are now reported on the calling channel, not the peer channel.
3) AMI DialEnd events have been altered to display the extension the call is
being forwarded to when relevant.
4) You can now get the values of channel variables for channels that are not
currently in the Stasis application. This brings the retrieval of channel
variables more in line with the rest of channel read operations since they
may be performed on channels not in Stasis.
ASTERISK-24134 #close
Reported by Matt Jordan
ASTERISK-24138 #close
Reported by Matt Jordan
Patches:
forward-shenanigans.diff uploaded by Matt Jordan (License #6283)
Review: https://reviewboard.asterisk.org/r/3899
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420794 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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420624 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
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
This patch adds a new module to Asterisk, res_hep_rtcp. The module subscribes
to the RTCP topics in Stasis and receives RTCP information back from the
message bus. It encodes into HEPv3 packets and sends the information to the
res_hep module for transmission.
Using this, someone with a Homer server can get live call quality monitoring
for all RTP-based channels in their Asterisk 12+ systems.
In addition, there were a few bugs in the RTP engine, res_rtp_asterisk, and
chan_pjsip that were uncovered by the tests written for the Asterisk Test
Suite. This patch fixes the following:
1) chan_pjsip failed to set its channel unique ids on its RTP instance on
outbound calls. It now does this in the appropriate location, in the
serialized call callback.
2) The rtp_engine was overflowing some values when packed into JSON.
Specifically, some longs and unsigned ints can't be be packed into integer
values, for obvious reasons. Since libjansson only supports integers,
floats, strings, booleans, and objects, we print these values into strings.
3) res_rtp_asterisk had a few problems:
(a) it would emit a source IP address of 0.0.0.0 if bound to that IP
address. We now use ast_find_ourip to get a better IP address, and
properly marshal the result into an ast_strdupa'd string.
(b) Reports can be generated with no report bodies. In particular, this
occurs when a sender is transmitting information to a receiver (who
will send no RTP back to the sender). As such, the sender has no report
body for what it received. We now properly handle this case, and the
sender will emit SR reports with no body. Likewise, if we receive an
RTCP packet with no report body, we will still generate the appropriate
events.
ASTERISK-24119 #close
........
Merged revisions 419823 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r419565 | mjordan | 2014-07-25 09:41:23 -0500 (Fri, 25 Jul 2014) | 21 lines
ARI: report duration values in LiveRecording objects
This patch adds three new fields to the LiveRecording model:
- total_duration: the total length of the live recording
- talking_duration: optional. The duration of talking energy that was
detected while the recording was made.
- silence_duration: optional. The duration of silence that was detected while
the recording was made.
These values are reported in the RecordingFinished ARI event.
When a DSP is enabled on the channel during the recording - which occurs when
the recording is created with max_silence_seconds (indicating that the user
actually cares about how much silence is in the file), we will report the
talking_duration and silence_duration in addition to the total_duration.
Review: https://reviewboard.asterisk.org/r/3770/
ASTERISK-24037 #close
Reported by: Samuel Galarneau
........
r419566 | mjordan | 2014-07-25 09:46:15 -0500 (Fri, 25 Jul 2014) | 1 line
Update CHANGES for r419565
........
Merged revisions 419565-419566 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The previous behavior was to simply set the accountcode of an outgoing
channel to the accountcode of the channel initiating the call. It was
done this way a long time ago to allow the accountcode set on the SIP/100
channel to be propagated to a local channel so the dialplan execution on
the Local;2 channel would have the SIP/100 accountcode available.
SIP/100 -> Local;1/Local;2 -> SIP/200
Propagating the SIP/100 accountcode to the local channels is very useful.
Without any dialplan manipulation, all channels in this call would have
the same accountcode.
Using dialplan, you can set a different accountcode on the SIP/200 channel
either by setting the accountcode on the Local;2 channel or by the Dial
application's b(pre-dial), M(macro) or U(gosub) options, or by the
FollowMe application's b(pre-dial) option, or by the Queue application's
macro or gosub options. Before Asterisk v12, the altered accountcode on
SIP/200 will remain until the local channels optimize out and the
accountcode would change to the SIP/100 accountcode.
Asterisk v1.8 attempted to add peeraccount support but ultimately had to
punt on the support. The peeraccount support was rendered useless because
of how the CDR code needed to unconditionally force the caller's
accountcode onto the peer channel's accountcode. The CEL events were thus
intentionally made to always use the channel's accountcode as the
peeraccount value.
With the arrival of Asterisk v12, the situation has improved somewhat so
peeraccount support can be made to work. Using the indicated example, the
the accountcode values become as follows when the peeraccount is set on
SIP/100 before calling SIP/200:
SIP/100 ---> Local;1 ---- Local;2 ---> SIP/200
acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200
peer: 200 /\ peer: 100 /\ peer: 200 /\ peer: 100
If a channel already has an accountcode it can only change by the
following explicit user actions:
1) A channel originate method that can specify an accountcode to use.
2) The calling channel propagating its non-empty peeraccount or its
non-empty accountcode if the peeraccount was empty to the outgoing
channel's accountcode before initiating the dial. e.g., Dial and
FollowMe. The exception to this propagation method is Queue. Queue will
only propagate peeraccounts this way only if the outgoing channel does not
have an accountcode.
3) Dialplan using CHANNEL(accountcode).
4) Dialplan using CHANNEL(peeraccount) on the other end of a local
channel pair.
If a channel does not have an accountcode it can get one from the
following places:
1) The channel driver's configuration at channel creation.
2) Explicit user action as already indicated.
3) Entering a basic or stasis-mixing bridge from a peer channel's
peeraccount value.
You can specify the accountcode for an outgoing channel by setting the
CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue
applications. Queue adds the wrinkle that it will not overwrite an
existing accountcode on the outgoing channel with the calling channels
values.
Accountcode and peeraccount values propagate to an outgoing channel before
dialing. Accountcodes also propagate when channels enter or leave a basic
or stasis-mixing bridge. The peeraccount value only makes sense for
mixing bridges with two channels; it is meaningless otherwise.
* Made peeraccount functional by changing accountcode propagation as
described above.
* Fixed CEL extracting the wrong ie value for the peeraccount. This was
done intentionally in Asterisk v1.8 when that version had to punt on
peeraccount.
* Fixed a few places dealing with accountcodes that were reading from
channels without the lock held.
AFS-65 #close
Review: https://reviewboard.asterisk.org/r/3601/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Say you wanted to include variables in an application map and have those
variables substituted and passed along to the application being executed;
currently this does not happen.
This patch adds this ability to pass channel variable values to an
application before being executed.
ASTERISK-22608 #close
Reported by: Michael L. Young
patches:
features_substitute_arguments_v2.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/3819/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
We have a new periodic beep feature but sometimes a user needs some sort of
feedback, without the need to have a periodic beep during the recording, to let
them know that MixMonitor started recording or ended the recording. The use
case where this patch is being used is when using Dynamic Features to start and
end MixMonitor.
This patch adds an option to play a beep when MixMonitor starts and an option to
play a beep when MixMonitor ends.
ASTERISK-24051 #close
Reported by: Michael L. Young
patches:
mixmonitor-play-beep-start-stop.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/3820/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new operation for stored recordings, copy. It takes an
existing stored recording and makes a copy of it in the same directory
or a relative directory under the stored recording directory.
/ari/recordings/stored/{recordingName}/copy?destinationRecordingName={copy_name}
This is particularly useful for voicemail-esque applications, which may need to
copy or move recordings around a directory structure.
Review: https://reviewboard.asterisk.org/r/3768/
ASTERISK-24036 #close
Reported by: Sam Galarneau
Tested by: Sam Galarneau
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Merged revisions 419021 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Whenever possible, audiohooks and framehooks will now be copied over
to the channel that the masquerading channel gets cloned into. This
should occur for all audiohooks and most framehooks. As a result,
in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now
deprecated and its behavior is essentially the new default for all
audiohooks, plus some additional audiohooks/framehooks.
Review: https://reviewboard.asterisk.org/r/3721/
........
Merged revisions 418914 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Most channel drivers let you specify a default accountcode to be set on
channels associated with a particular peer/endpoint/object. Prior to this
patch, chan_pjsip/res_pjsip did not support such a setting.
This patch adds a new setting to the res_pjsip endpoint object, 'accountcode'.
When a channel is created that is associated with an endpoint with this value
set, the channel will automatically have its accountcode property set to the
value configured for the endpoint.
Review: https://reviewboard.asterisk.org/r/3724/
ASTERISK-24000 #close
Reported by: Matt Jordan
........
Merged revisions 418756 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds support for the PostgreSQL application_name connection setting.
When the appropriate PostgreSQL module's configuration is set with an
application name, the name will be passed to PostgreSQL on connection and
displayed in the database's pg_stat_activity view, as well as in CSV logs. This
aids in managing which applications/servers are connected to a PostgreSQL
database, as well as tracing the activity of those connections.
Review: https://reviewboard.asterisk.org/r/3591
ASTERISK-23737 #close
Reported by: Gergely Domodi
patches:
pgsql_application_name.patch uploaded by Gergely Domodi (License 6610)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Billing records are fair,
To get paid is quite bright,
You should really use ODBC;
Good-bye cdr_sqlite.
Microsoft did once push H.323,
Hell, we all remember NetMeeting.
But try to compile chan_h323 now
And you will take quite a beating.
The XMPP and SIP war was fierce,
And in the distant fray
Was birthed res_jabber/chan_jingle;
But neither to stay.
For everyone did care and chase what Google professed.
"Free Internet Calling" was what devotees cried,
But Google did change the specs so often
That the developers were happy the day chan_gtalk died.
And then there was that odd application
Dedicated to the Polish tongue.
app_saycountpl was subsumed by Say;
One could say its bell was rung.
To read and parse a file from the dialplan
You could (I guess) use an application.
app_readfile did fill that purpose, but I think
A function is perhaps better in its creation.
Barging is rude, I'm not sure why we do it.
Inwardly, the caller will probably sigh.
But if you really must do it,
Don't use app_dahdibarge, use ChanSpy.
We all despise the sound of tinny robots
It makes our queues so cold.
To control such an abomination
It's better to not use Wait/SetMusicOnHold.
It's often nice to know properties of a channel
It makes our calls right
We have a nice function called CHANNEL
And so SIPCHANINFO is sent off into the night.
And now things get odd;
Apparently one could delimit with a colon
Properties from the SIPPEER function!
Commas are in; all others are done.
Finally, a word on pipes and commas.
We're sorry. We can't say it enough.
But those compatibility options in asterisk.conf;
To maintain them forever was just too tough.
This patch removes:
* cdr_sqlite
* chan_gtalk
* chan_jingle
* chan_h323
* res_jabber
* app_saycountpl
* app_readfile
* app_dahdibarge
It removes the following applications/functions:
* WaitMusicOnHold
* SetMusicOnHold
* SIPCHANINFO
It removes the colon delimiter from the SIPPEER function.
Finally, it also removes all compatibility options that were configurable from
asterisk.conf, as these all applied to compatibility with Asterisk 1.4 systems.
Review: https://reviewboard.asterisk.org/r/3698/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch enables Perfect Forward Secrecy (PFS) in Asterisk's core TLS API.
Modules that wish to enable PFS should consider the following:
- Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
specify a ECDHE cipher suite in a module's configuration, for example:
tlscipher=AES128-SHA:DES-CBC3-SHA
- Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
into the private key file, i.e., tlsprivatekey. For an example, see the
default dh2048.pem at
http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
- Because clients expect the server to prefer PFS, and because OpenSSL sorts
its cipher suites by bit strength, (see "openssl ciphers -v DEFAULT")
consider re-ordering your cipher suites in the conf file. For example:
tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
will use PFS when offered by the client. Clients which do not offer PFS
fall-back to AES-128 (or even 3DES as recommend by RFC 3261).
Review: https://reviewboard.asterisk.org/r/3647/
ASTERISK-23905 #close
Reported by: Alexander Traud
patches:
tlsPFS_for_HEAD.patch uploaded by Alexander Traud (License 6520)
tlsPFS.patch uploaded by Alexander Traud (License 6520)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds support for the Japanese language to both the say family of
applications, as well as for VoiceMail and VoiceMailMain. A new pack of
language sounds will be released at the same time as the next major version
of Asterisk to support the new language features.
The language features can be enabled using a language code of 'ja'.
Review: https://reviewboard.asterisk.org/r/3477
ASTERISK-23324 #close
Reported by: Kevin McCoy
patches:
app_voicemail.c.20140226.jb.patch uploaded by Kevin McCoy (License 6586)
say.c.20140226.jb.patch uploaded by Kevin McCoy (License 6586)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch enables the jack-audiohook to cope with dynamic sampling rates from
and to Asterisk. Information from the channel is taken to derive the channel's
sampling rate, suiting SLINxx format and frame->datalen.
There are stil a few limitations after this patch:
* Required information is taken from the channel during initialization as
the audiohook does not provide this information.
Audiohook.internal_sampl_rate(...) is set later, but no callback is available
to inform app_jack.
* Frame.datalen is computed using "rate / 50" assuming a ptime of 20ms.
There is no internal API available to determine datalen for a SLINxx.
* Ringbuffer size is now dynamic depending on the value of frame.datalen
(see above) and the number of frames, which are in RINGBUFFER_FRAME_CAPACITY,
that need to fit.
Review: https://reviewboard.asterisk.org/r/3618
Note that the patch being committed here is based on the patch posted on
ASTERISK-23836. However, Matthis Schmieder also provided a patch to enable
this functionality, and that patch is noted below.
ASTERISK-20696 #close
Reported by: Matthis Schmieder
patches:
app_jack.patch uploaded by Matthis Schmieder (License 6445)
ASTERISK-23836 #close
Reported by: Dennis Guse
patches:
patch-app_jack.c uploaded by Dennis Guse (License 6513)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* SS7 support now requires libss7 v2.0 or later. The new libss7 is not
backwards compatible.
* Added SS7 support for connected line and redirecting.
* Most SS7 CLI commands are reworked as well as new SS7 commands added.
See online CLI help.
* Added several SS7 config option parameters described in
chan_dahdi.conf.sample.
* ISUP timer support reworked and now requires explicit configuration.
See ss7.timers.sample.
Special thanks to Kaloyan Kovachev for his support and persistence in
getting the original patch by adomjan updated and ready for release.
SS7-27 #close
Reported by: adomjan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new channel function TALK_DETECT that, when set on a
channel, causes events indicating the start/stop of talking on a channel to be
emitted to both AMI and ARI clients.
The function allows setting both the silence threshold (the length of silence
after which we decide no one is talking) as well as the talking threshold (the
amount of energy that counts as talking). Parameters can be updated on a channel
after talk detection has been enabled, and talk detection can be removed at
any time.
The events raised by the function use a nomenclature similar to existing AMI/ARI
events.
For AMI: ChannelTalkingStart/ChannelTalkingStop
For ARI: ChannelTalkingStarted/ChannelTalkingFinished
Review: https://reviewboard.asterisk.org/r/3563/
#ASTERISK-23786 #close
Reported by: Matt Jordan
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Improvements to the agent pool functionality.
* AgentRequest no longer hangs up the caller if the agent fails to connect
with the caller. It now continues in the dialplan.
* AgentRequest returns AGENT_STATUS set to NOT_CONNECTED if the agent
failed to connect with the call. Most likely because the agent did not
acknowledge the call in time or got disconnected.
* The agent alerting play file configured by the agent.conf custom_beep
option can now be disabled by setting the option to an empty string. The
agent is effectively alerted to a call presence when MOH stops.
* Fixed bridge reference leak when the agent connects with a caller.
ASTERISK-23499 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3551/
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User events can now be generated from ARI. Events can be signalled with
arbitrary json variables, and include one or more of channel, bridge, or
endpoint snapshots. An application must be specified which will receive
the event message (other applications can subscribe to it). The message
will also be delivered via AMI provided a channel is attached. Dialplan
generated user event messages are still transmitted via the channel, and
will only be received by a stasis application they are attached to or if
the channel is subscribed to.
This change also introduces the multi object blob mechanism used to send
multiple snapshot types in a single message. The dialplan app UserEvent
was also changed to use multi object blob, and a new stasis message type
created to handle them.
ASTERISK-22697 #close
Review: https://reviewboard.asterisk.org/r/3494/
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In r411189, some behavior was changed which made sendrpid behavior
act in a more trusting manner by sending full user data for peers
set with private caller presence in P-Asserted-Identity headers.
Since this changed long time expected behaviors, we decided to pull
that patch when that was pointed out by the community. Instead, this
patch provides a trust_id_outbound setting which will expose the data
per RFC-3325 if set to 'yes' and simply not send the PAI/RPID headers
at all if set to 'no'. By default trust_id_outbound will be set to
'legacy' which will preserve the behavior prior to these patches.
Extra special thanks to Walter Doekes for providing advice and
feedback.
(closes issue AST-1301)
(closes issue ASTERISK-19465)
Reported by: Krzysztof Chmielewski
Review: https://reviewboard.asterisk.org/r/3447/
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Adds a tones URI type to the playback resource. The tone can be specified by
name (from indications.conf) or by a tone pattern. In addition, tonezone can
be specified in the URI (by appending ;tonezone=<zone>). Tones must be
stopped manually in order for a stasis control to move on from playback of
the tone. Tones may be paused, resumed, restarted, and stopped. They may
not be rewound or fast forwarded (tones can't be controlled in a way that
lets you skip around from note to note and pausing and resuming will also
restart the tone from the beginning). Tests are currently in development
for this feature (https://reviewboard.asterisk.org/r/3428/).
(closes issue ASTERISK-23433)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3427/
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This patch is a continuation of https://reviewboard.asterisk.org/r/3349/,
committed in r412303.
It resolves a finding oej had that the phone-context be available in a
channel variable separate from SIPDOMAIN. This patch adds that variable as
SIPURIPHONECONTEXT. It also allows a local number (or global number specified
in the TEL URI) to be used to look up as a peer.
(issue ASTERISK-17179)
Review: https://reviewboard.asterisk.org/r/3349/
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Add an option to enable a periodic beep to be played into a call if it
is being recorded. If enabled, it uses the PERIODIC_HOOK() function
internally to play the 'beep' prompt into the call at a specified
interval. This option is provided for both Monitor() and
MixMonitor().
Review: https://reviewboard.asterisk.org/r/3424/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds support for handling TEL URIs in inbound INVITE requests.
This includes the Request URI and the From URI. The number specified in
the Request URI will be the destination of the inbound channel in the dialplan.
The phone-context specified in the Request URI will be stored in the
TELPHONECONTEXT channel variable.
Review: https://reviewboard.asterisk.org/r/3349
ASTERISK-17179 #close
Reported by: Geert Van Pamel
Tested by: Geert Van Pamel
patches:
asterisk-12.0.0-chan_sip-RFC3966_patch.txt uploaded by Geert Van Pamel (License 6140)
asterisk-12.0.0-reqresp_parser-RFC3966_patch.txt uploaded by Geert Van Pamel (License 6140)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes a parsing error that occurred during the processing of
the AMI action. The error did not result in MixMonitor itself
misbehaving, but it could result in the AMI response not giving
correct information back.
The new header allows for one to specify a post-process command
to run when recording finishes. Previously, in order to do this,
the post-process command would have to be placed at the end of
the Options: header.
Patches: mixmonitor_command_2.patch by jhardin (License #6512)
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This commit introduces a new dialplan function, PERIODIC_HOOK().
It allows you run to a dialplan hook on a channel periodically. The
original use case that inspired this was the ability to play a beep
periodically into a call being recorded. The implementation is much
more generic though and could be used for many other things.
The implementation makes heavy use of existing Asterisk components.
It uses a combination of Local channels and ChanSpy() to run some
custom dialplan and inject any audio it generates into an active call.
The other important bit of the implementation is how it figures out
when to trigger the beep playback. This implementation uses the
audiohook API, even though it's not actually touching the audio in any
way. It's a convenient way to get a callback and check if it's time
to kick off another beep. It would be nice if this was timer event
based instead of polling based, but unfortunately I don't see a way to
do it that won't interfere with other things.
Review: https://reviewboard.asterisk.org/r/3362/
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This patch adds the following:
(1) A new module, res_hep, which implements a generic packet capture agent for
the Homer Encapsulation Protocol (HEP) version 3. Note that this code is based
on a patch provided by Alexandr Dubovikov; I basically just wrapped it up,
added configuration via the configuration framework, and threw in a
taskprocessor.
(2) A new module, res_hep_pjsip, which forwards all SIP message traffic that
passes through the res_pjsip stack over to res_hep for encapsulation and
transmission to a HEPv3 capture server.
Much thanks to Alexandr for his Asterisk patch for this code and for a *lot*
of patience waiting for me to port it to 12/trunk. Due to some dithering on
my part, this has taken the better part of a year to port forward (I still
blame CDRs for the delay).
ASTERISK-23557 #close
Review: https://reviewboard.asterisk.org/r/3207/
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This patch does the following:
* It updates the AMI version to 2.2.0 to indicate backwards compatible
changes have been made since the last release
* It updates the ARI version to 1.2.0 to indicate backwards compatible
changes have been made since the last release
* It updates the UPGRADE/CHANGES files with changes that were not
mentioned
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This change turns the bridge type field into a comma separated list of attributes.
These attributes include: mixing, holding, dtmf_events, and proxy_media. By setting
the various attributes a user can control the type of bridge created with the
behavior they need for their application.
(closes issue ASTERISK-23437)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3359/
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This change enables DNS client support within PJSIP. System
nameservers are automatically discovered using res_init or
res_ninit. If this fails then PJSIP will resort to using
gethostbyname for resolution.
By enabling this support we gain SRV support, failover, and
weight support.
(closes issue ASTERISK-23435)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3343/
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This patch creates the AST_SORCERY dialplan function which allows someone to
retrieve any value from a sorcery-based config file. It's similar to
AST_CONFIG.
The creation of the function itself was fairly straightforward but it required
changes to the underlying sorcery infrastructure that rippled into individual
sorcery objects. The changes stemmed from inconsistencies in how sorcery
created ast_variable objectsets from sorcery objects and the inconsistency
in how individual objects used that feature especially when it came to
parameters that can be specified multiple times like contact in aor and match
in identify. You can read more here...
http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html
So, what this patch does, besides actually creating the AST_SORCERY function,
is the following...
* Creates ast_variable_list_append which is a helper to append one ast_variable
list to another.
* Modifies the ast_sorcery_object_field_register functions to accept the
already-defined sorcery_fields_handler callback.
* Modifies ast_sorcery_objectset_create to accept a parameter indicating return
type preference...a single ast_variable with all values concatenated or an
ast_variable list with multiple entries. Also fixed a few bugs.
* Modifies individual sorcery object implementations to use the new function
definition of the ast_sorcery_object_field_register functions.
* Modifies location.c and res_pjsip_endpoint_identifier_ip.c to implement
sorcery_fields_handler handlers so they return multiple occurrences as an
ast_variable_list.
* Added a whole bunch of tests to test_sorcery.
(closes issue ASTERISK-22537)
Review: http://reviewboard.asterisk.org/r/3254/
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The changes log was written with language that was a little too internal
Asterisk specific, so it's been changed to be more in the frame of reference
of an ARI user. Also, previously the AMI event changes were omitted from the
change log as well as the ability to include a bridge name in the ARI post
bridges command.
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Added a note in the changes file about the new 'StatusText' field that was
added to the 'ExtensionStatus' event.
(issue ASTERISK-23154)
Reported by: Jonathan Rose
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This change is in preparation for external MWI support.
Removed code from the system for normal mailbox handling that appends
@default to the mailbox identifier if it does not have a context. The
only exception is the legacy hasvoicemail users.conf option. The legacy
option will only work for app_voicemail mailboxes. The system cannot make
any assumptions about the format of the mailbox identifer used by
app_voicemail.
chan_sip and chan_dahdi/sig_pri had the most changes because they both
tried to interpret the mailbox identifier. chan_sip just stored and
compared the two components. chan_dahdi actually used the box
information.
The ISDN MWI support configuration options had to be reworked because
chan_dahdi was parsing the box@context format to get the box number. As a
result the mwi_vm_boxes chan_dahdi.conf option was added and is documented
in the chan_dahdi.conf.sample file.
Review: https://reviewboard.asterisk.org/r/3072/
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When doing the rework of the CDR engine that pushed all of the logic into cdr.c
and made it respond to changes in channel state over Stasis, we knew that
accessing the CDR engine from the dialplan would be "slightly"
non-deterministic. Dialplan threads would be accessing CDRs while Stasis
threads would be updating the state of said CDRs - whereas in the past,
everything happened on the dialplan threads. Tests have shown that "slightly"
is in reality "very".
This patch synchronizes things by making the dialplan applications/functions
that manipulate CDRs do so over Stasis. ForkCDR, NoCDR, ResetCDR, CDR, and
CDR_PROP now all use Stasis to send their requests over to the CDR engine,
and synchronize on the channel Stasis topic via a subscription so that they
return their values/control to the dialplan at the appropriate time.
While going through this, the following changes were also made:
* DISA, which can reset the CDR when a user successfully authenticates, now
just uses the ResetCDR app to do this. This prevents having to duplicate
the same Stasis synchronization logic in that application.
* Answer no longer disables CDRs. It actually didn't work anyway - calling
DISABLE on the channel's CDR doesn't stop the CDR from getting the Answer
time - it just kills all CDRs on that channel, which isn't what the caller
would intend.
(closes issue ASTERISK-22884)
(closes issue ASTERISK-22886)
Review: https://reviewboard.asterisk.org/r/3057/
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For the time, this is only useful for retrieving the filename.
The purpose of this function is to better facilitate multiple
mixmonitors per channel. Setting a MIXMONITOR_FILENAME channel
variable is not conducive to such behavior, so allowing finer
grained access to individual mixmonitor properties improves
the situation. The MIXMONITOR_FILENAME channel variable is still
set, though, so there is no worry about backwards compatibility.
Review: https://reviewboard.asterisk.org/r/3023
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* Made PickupChan() search by channel uniqueids if the search could not
find a channel by name.
* Ensured PickupChan() never considers the picking channel for pickup.
* Made PickupChan() option p use a common search by name routine. The
original search was erroneously case sensitive.
(issue AFS-42)
Review: https://reviewboard.asterisk.org/r/3017/
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By the time the directory application exits, a channel variable
DIRECTORY_RESULT will be set for the channel that invoked it which
can be used to determine the reason for exit. The changes log and
the app_directory documentation contain specific details about
each of the possible values for DIRECTORY_RESULT.
Review: https://reviewboard.asterisk.org/r/3016/
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Similar to how background works, if a say application is called with
this variable set to 'true', 'yes', 'on', etc. then using DTMF while
the say action is in progress will result in the channel jumping to
that extension in the dialplan.
Review: https://reviewboard.asterisk.org/r/3011/
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Also adds the ability to clear all profile items and makes behavior more
consistent with documentation as when choosing whether to use CONFBRIDGE
datastore profiles or the application arguments to the confbridge application.
(closes issue ASTERISK-22760)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2971/
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ConfBridge now has the ability to set the language of announcements to the
conference. The language can be set on a bridge profile in
confbridge.conf or by the dialplan function
CONFBRIDGE(bridge,language)=en.
(closes issue ASTERISK-19983)
Reported by: Jonathan White
Patches:
M19983_rev2.diff (license #5138) patch uploaded by junky (modified)
Tested by: rmudgett
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With the new work in Asterisk 12, there are some uses of the
optional_api that are prone to failure. The details are rather involved,
and captured on [the wiki][1].
This patch addresses the issue by removing almost all of the magic from
the optional API implementation. Instead of relying on weak symbol
resolution, a new optional_api.c module was added to Asterisk core.
For modules providing an optional API, the pointer to the implementation
function is registered with the core. For modules that use an optional
API, a pointer to a stub function, along with a optional_ref function
pointer are registered with the core. The optional_ref function pointers
is set to the implementation function when it's provided, or the stub
function when it's now.
Since the implementation no longer relies on magic, it is now supported
on all platforms. In the spirit of choice, an OPTIONAL_API flag was
added, so we can disable the optional_api if needed (maybe it's buggy on
some bizarre platform I haven't tested on)
The AST_OPTIONAL_API*() macros themselves remained unchanged, so
existing code could remain unchanged. But to help with debugging the
optional_api, the patch limits the #include of optional API's to just
the modules using the API. This also reduces resource waste maintaining
optional_ref pointers that aren't used.
Other changes made as a part of this patch:
* The stubs for http_websocket that wrap system calls set errno to
ENOSYS.
* res_http_websocket now properly increments module use count.
* In loader.c, the while() wrappers around dlclose() were removed. The
while(!dlclose()) is actually an anti-pattern, which can lead to
infinite loops if the module you're attempting to unload exports a
symbol that was directly linked to.
* The special handling of nonoptreq on systems without weak symbol
support was removed, since we no longer rely on weak symbols for
optional_api.
[1]: https://wiki.asterisk.org/wiki/x/wACUAQ
(closes issue ASTERISK-22296)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2797/
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The patch from ASTERISK-21965 was committed perhaps a bit too hastily. Walter
and Tzafrir have pointed out numerous issues with the approach and have
propsed an alternative in r/2757. Since it's not a time critical issue and
is not worth holding up the release of 12 for it, I've gone ahead and reverted
r394939 from 12/trunk and re-opened ASTERISK-21965.
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This patch replaces contrib/realtime/ with a new setup for managing the
database schema required for database integration with Asterisk. In
addition to initializing a database with the proper schema, alembic can do a
database migration to assist with upgrading Asterisk in the future.
Hopefully this helps make setting up and operating Asterisk with a database
easier.
With this the schema only needs to be maintained in one place instead of
once per database. The schemas I have added here have a bit of improvement
over the examples that were there before (some added consistency and added
some missing indexes). Managing the schema in one place here also applies
to all databases supported by SQLAlchemy.
See contrib/ast-db-manage/README.md for more details.
Review: https://reviewboard.asterisk.org/r/2731
patch by Russell Bryant (license 6300)
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This adds a new dialplan application, SayAlphaCase, that performs much
the same function as SayAlpha except that it takes additional options
which allow the user to specify whether the case of each letter should
be announced for uppercase, lowercase, or all letters. Similar
functionality has been added to the SAY ALPHA AGI command via an
optional parameter.
Original Patch by: Kevin Scott Adams
Reported by: Kevin Scott Adams
Review: https://reviewboard.asterisk.org/r/2725/
(closes issue ASTERISK-20782)
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This essentially makes app_queue usable again. From reviewboard:
* Reporting of transfers and call completion is done by creating stasis
subscriptions and listening for specific events in order to determine
when the call is finished (either via a transfer or hangup).
* Dial end messages have been added where they were previously missing.
* Queue stats are properly being updated again once calls have finished.
* AgentComplete stasis messages and AMI events are now occurring again.
* Mixmonitor starting has been factored into its own function and uses the
Mixmonitor API now instead of using ast_pbx_run()
In addition to the changes in app_queue, there are several supplementary changes as well:
* Queue logging now differentiates between attended and blind transfers. A
note about this is in the CHANGES file.
* Local channel optimization events now report more information. This
includes which of the two local channels involved is the destination of
the optimization, the channel that is replacing the destination local channel,
and an identifier so that begin and end events can be matched to each other.
The end events are now sent whether the optimization was successful or not and
includes an indicator of whether the optimization was successful.
* Changes were made to features and bridging_basic so that additional flags may
be set on a bridge. This is necessary because the queue requires that its
bridge only allows move-swap local channel optimizations into the bridge.
(closes issue ASTERISK-21517)
Reported by Matt Jordan
(closes issue ASTERISK-21943)
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2694
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The SIP_CODEC family of variables let you set the preferred codec to be
offered on an outbound INVITE request. However, for video calls, you need to
be able to set both the audio and video codecs to be offered. This patch lets
the SIP_CODEC variables accept a comma delineated list of codecs. The first
codec in the list is set as the preferred codec; additional codecs are still
offered however.
This lets a dialplan writer set both audio and video codecs, e.g.,
Set(SIP_CODEC=ulaw,h264)
Note that this feature was written by both Dennis Guse and Frank Haase
Review: https://reviewboard.asterisk.org/r/2728
(closes issue ASTERISK-21976)
Reported by: Denis Guse
Tested by: mjordan, sysreq
patches:
patch-channels-chan__sip.c-393919 uploaded by dennis.guse (license 6513)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It moves the pickup code out of features.c and into pickup.c
* It removes the vast majority of dead code out of features.c. In particular,
this includes the parking code.
(issue ASTERISK-22134)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the ability in Queue to raise a hint when a member's paused
state changes. The hint uses the form 'Queue:{queue_name}_pause_{member_name}',
where {queue_name} and {member_name} are the name of the queue and the name
of the member to subscribe to, respectively.
For example: exten => 8501,hint,Queue:sales_pause_mark.
Members will show as In Use when paused.
Note that the format of the queue pause hint was changed slightly from what
is on the issue to accomodate suggestion on the code review.
Review: https://reviewboard.asterisk.org/r/2254
(closes issue ASTERISK-20842)
Reported by: Philippe Lindheimer
patches:
qpause-10-378206.diff uploaded by Philippe Lindheimer (license 5519)
qpause-11-378206.diff uploaded by Philippe Lindheimer (license 5519)
qpause-trunk-378206.diff uploaded by Philippe Lindheimer (license 5519)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter sets
with a given prefix or defaulting to no prefix. This also prevents
documentation from becoming fractured and out of date by keeping all
variations of the documentation in template form such that it only
needs to be updated once and keeps maintenance to a minimum.
Review: https://reviewboard.asterisk.org/r/2708/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch allows starting playback of audio through the CONTROL STREAM FILE
AGI command to start at a particular offset. It will also return the final
position of the file in the 'endpos' attribute.
(closes issue ASTERISK-17803)
Reported by: Murray Melvin
patches:
res_agi.c.r316293.diff uploaded by murraytm (license 6221)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch modifies the behavior of safe_asterisk in two ways:
(1) It modifies the Asterisk Makefile such that safe_asterisk is always
installed on a 'make install'. This was done as bugfixes in the
safe_asterisk script were not applied in previous version of Asterisk
without first removing the old version of the script.
(2) In order to keep a newly installed version of safe_asterisk from impacting
local modifications, a new config file - safe_asterisk.conf.sample - has
been provided. Settings that were previously modified in safe_asterisk can
be set there instead.
(closes issue ASTERISK-21965)
Reported by: Jeremy Kister
patches:
safe_asterisk.patch uploaded by jkister (License 6232)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch modifies manager to allow the allowmultiplelogin setting to be set
on an account by account basis. When set in the general context, it will act
as the default for the defined accounts. Setting it in the account will
override the general setting.
(closes issue ASTERISK-21324)
Reported by: vldmr
patches:
asterisk-manager-per-user-allowmultiplelogin.patch uploaded by vldmr (License 6487)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a new CEL event type, AST_CEL_LOCAL_OPTIMIZE, to represent
local channel optimizations. Local channel optimizations were one of
several things conveyed by the now defunct BRIDGE_UPDATE event type.
This also adds a unit test to test generation of this new CEL event.
Review: https://reviewboard.asterisk.org/r/2676/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds CEL support for blind and attended transfers and call pickup.
During the course of adding this functionality I noticed that
CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are particularly
useless without a bridge identifier, so I added that as well.
This adds tests for blind transfers, several types of attended
transfers, and call pickup.
The extra field in CEL records now consists of a JSON blob whose fields
are defined on a per-event basis.
Review: https://reviewboard.asterisk.org/r/2658/
(closes issue ASTERISK-21565)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the ability to kick all users out of a conference from the
ConfBridge kick CLI command. It is invoked by passing 'all' as the channel
parameter to the CLI command, i.e., "confbridge kick <conf> all".
Note that this patch was modified slightly to conform to trunk.
(closes issue ASTERISK-21827)
Reported by: dorianlogan
patches:
kickall-patch_v2.diff uploaded by dorianlogan (License 6504)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ill conceived chan_agent is no more. It is now replaced by
app_agent_pool.
Agents login using the AgentLogin() application as before. The
AgentLogin() application no longer does any authentication.
Authentication is now the responsibility of the dialplan. (Besides, the
authentication done by chan_agent did not match what the voice prompts
asked for.)
Sample extensions.conf
[login]
; Sample agent 1001 login
; Set COLP for in between calls so the agent does not see the last caller COLP.
exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>)
; Give the agent DTMF transfer and disconnect features when connected to a caller.
same => n,Set(CHANNEL(dtmf-features)=TX)
same => n,AgentLogin(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()
[caller]
; Sample caller direct connect to agent 1001
exten => 800,1,AgentRequest(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()
; Sample caller going through a Queue to agent 1001
exten => 900,1,Queue(agent_q)
same => n,Hangup()
Sample queues.conf
[agent_q]
member => Local/800@caller,,SuperAgent,Agent:1001
Under the hood operation overview:
1) Logged in agents wait for callers in an agents holding bridge.
2) Caller requests an agent using AgentRequest()
3) A basic bridge is created, the agent is notified, and caller joins the
basic bridge to wait for the agent.
4) The agent is either automatically connected to the caller or must ack
the call to connect.
5) The agent is moved from the agents holding bridge to the basic bridge.
6) The agent and caller talk.
7) The connection is ended by either party.
8) The agent goes back to the agents holding bridge.
To avoid some locking issues with the agent holding bridge, I needed to
make some changes to the after bridge callback support. The after bridge
callback is now a list of requested callbacks with the last to be added
the only active callback. The after bridge callback for failed callbacks
will always happen in the channel thread when the channel leaves the
bridging system or is destroyed.
(closes issue ASTERISK-21554)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2657/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It adds a virtual table of callbacks to core_unreal. These callbacks can be
supplied by concrete implementations of "unreal" channel drivers, which lets
the unreal channel driver call specific functionality when it performs some
action. Currently, this is done to notify implementations when an
optimization operation has begun, and when an optimization operation has
succeeded.
* It adds Stasis-Core messages for Local channel bridging and Local channel
optimization. Local channel optimization is now two events: a Begin and an
End. Some consumers of Stasis-Core may want to know when an operation is
beginning so that they can 'prepare' their information; others will be more
concerned about when the operation has completed, so that they can 'fix up'
information. Stasis-Core allows for both, as does AMI.
Review: https://reviewboard.asterisk.org/r/2552
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It adds a new soft hangup flag AST_SOFTHANGUP_HANGUP_EXEC that is set when a
channel is executing dialplan hangup logic, i.e., the 'h' extension or a
hangup handler. Stasis messages now also convey the soft hangup flag so
consumers of the messages can know when a channel is executing said
hangup logic.
* It adds a new channel flag, AST_FLAG_DEAD, which is set when a channel is
well and truly dead. Not just a zombie, but dead, Jim. Manager, CEL, CDRs,
and other consumers of Stasis have been updated to look for this flag to
know when the channel should by lying six feet under.
* The CDR engine has been updated to better handle a channel entering and
leaving a bridge. Previously, a new CDR was automatically created when a
channel left a bridge and put into the 'Pending' state; however, this
way of handling CDRs made it difficult for the 'endbeforehexten' logic to
work correctly - there was always a new CDR waiting in the hangup logic
and, even if 'ended', wouldn't be the CDR people wanted to inspect in the
hangup routine. This patch completely removes the Pending state and instead
defers creation of the new CDR until it gets a new message that requires
a new CDR.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This process also involved a large amount of rework regarding how to redial
the Parker when a channel leaves a parking lot due to timeout. An attended
transfer channel variable has been added to attended transfers to extensions
that will eventually park (but haven't at the time of transfer) as well.
This resolves one of the two BUGBUG comments remaining in res_parking.
(issues ASTERISK-21877)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2638/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393704 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In addition to porting those features, they now enjoy greater feature parity
with one another. Specifically, AutoMixMon now has a start and stop
message that can be specified with TOUCH_MIXMONITOR_MESSAGE_START and
TOUCH_MIXMONITOR_MESSAGE_STOP.
(closes issue ASTERISK-21553)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2620/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The patch for CDRs moved around a lot of content in CHANGES to try and
organize the areas that were affected. This missed some changes that went
in with a merge and removed some updates - this patch adds them back in.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is the initial push to update Asterisk's CDR engine for the new
bridging framework. This patch guts the existing CDR engine and builds the new
on top of messages coming across Stasis. As changes in channel state and bridge
state are detected, CDRs are built and dispatched accordingly. This
fundamentally changes CDRs in a few ways.
(1) CDRs are now *very* reflective of the actual state of channels and bridges.
This means CDRs track well with what an actual channel is doing - which
is useful in transfer scenarios (which were previously difficult to pin
down). It does, however, mean that CDRs cannot be 'fooled'. Previous
behavior in Asterisk allowed for CDR applications, channels, and other
properties to be spoofed in parts of the code - this no longer works.
(2) CDRs have defined behavior in multi-party scenarios. This behavior will not
be what everyone wants, but it is a defined behavior and as such, it is
predictable.
(3) The CDR manipulation functions and applications have been overhauled. Major
changes have been made to ResetCDR and ForkCDR in particular. Many of the
options for these two applications no longer made any sense with the new
framework and the (slightly) more immutable nature of CDRs.
There are a plethora of other changes. For a full description of CDR behavior,
see the CDR specification on the Asterisk wiki.
(closes issue ASTERISK-21196)
Review: https://reviewboard.asterisk.org/r/2486/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer
channel driver specific. If the channel variable is set on the
transferrer channel, the sound will be played to the target of an attended
transfer.
* The channel variable BRIDGEPEER becomes a comma separated list of peers
in a multi-party bridge. The BRIDGEPEER value can have a maximum of 10
peers listed. Any more peers in the bridge will not be included in the
list. BRIDGEPEER is not valid in holding bridges like parking since those
channels do not talk to each other even though they are in a bridge.
* The channel variable BRIDGEPVTCALLID is only valid for two party bridges
and will contain a value if the BRIDGEPEER's channel driver supports it.
* The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and
is removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name
that activated the dynamic feature.
* The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are
set only on the channel executing the dynamic feature. Executing a
dynamic feature on the bridge peer in a multi-party bridge will execute it
on all peers of the activating channel.
(closes issue ASTERISK-21555)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2582/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The attended transfer API call can complete the attended transfer in a number of ways
depending on the current bridged states of the channels involved.
The hiding of masquerades is done in some bridging-related functions, such as the manager
Bridge action and the Bridge dialplan application. In addition, call pickup was edited
to "move" a channel rather than masquerade it.
Review: https://reviewboard.asterisk.org/r/2511
(closes issue ASTERISK-21334)
Reported by Matt Jordan
(closes issue Asterisk-21336)
Reported by Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch moves a number of AMI events over to the Stasis-Core message bus.
This includes:
* ChanSpyStart/Stop
* MonitorStart/Stop
* MusicOnHoldStart/Stop
* FullyBooted/Reload
* All Voicemail/MWI related events
In addition, it adds some Stasis-Core and AMI support for generic AMI messages,
refactors the message router in AMI to use a single router with topic
forwarding for the topics that AMI cares about, and refactors MWI message
types and topics to be more name compliant.
Review: https://reviewboard.asterisk.org/r/2532
(closes issue ASTERISK-21462)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Breaks many things until they can be reworked. A partial list:
chan_agent
chan_dahdi, chan_misdn, chan_iax2 native bridging
app_queue
COLP updates
DTMF attended transfers
Protocol attended transfers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change adds a framework in res_stasis for handling events from
channel topics. JSON event generation and validation code is created
from event documentation in rest-api/api-docs/events.json to assist in
JSON event generation, ensure consistency, and ensure that accurate
documentation is available for ALL events that are received by
res_stasis applications.
The userevent application has been refactored along with the code that
handles userevent channel blob events to pass the headers as key/value
pairs in the JSON blob. As a side-effect, app_userevent now handles
duplicate keys by overwriting the previous value.
Review: https://reviewboard.asterisk.org/r/2428/
(closes issue ASTERISK-21180)
Patch-By: Kinsey Moore <kmoore@digium.com>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change does the following:
1. Adds the sorcery realtime module
2. Adds unit tests for the sorcery realtime module
3. Changes the realtime core to use an ast_variable list instead of variadic arguments
4. Changes all realtime drivers to accept an ast_variable list
Review: https://reviewboard.asterisk.org/r/2424/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The settings saved on the channel for FEATURE()/FEATUREMAP() were only
for that channel. This patch adds the ability to have these settings
inherited to child channels if you set FEATURE(inherit)=yes.
Closes issue ASTERISK-21306.
Review: https://reviewboard.asterisk.org/r/2415/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* A new Stasis payload has been defined for multi-channel messages. This
payload can store multiple ast_channel_snapshot objects along with a single
JSON blob. The payload object itself is opaque; the snapshots are stored
in a container keyed by roles. APIs have been provided to query for and
retrieve the snapshots from the payload object.
* The Dial AMI events have been refactored onto Stasis. This includes dial
messages in app_dial, as well as the core dialing framework. The AMI events
have been modified to send out a DialBegin/DialEnd events, as opposed to
the subevent type that was previously used.
* Stasis messages, types, and other objects related to channels have been
placed in their own file, stasis_channels. Unit tests for some of these
objects/messages have also been written.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
HangupRequest and SoftHangupRequest are now ast_channel_blob Stasis
messages, with the cause code as an optional field in the blob.
NewCallerid now simply watches for changes in the callerid information
in channel snapshots, and creates the AMI event appropriately.
Since the original NewCallerid event honored the channelvars setting
in manager.conf, the channel variables configured there had to become
a part of the channel snapshot. These are now a part of every snapshot
based event, making the configuration description "every time a
channel-oriented event is emitted" less of a lie.
There a a few other changes wrapped up in here as well.
* When ast_channel_topic() is given NULL for a channel, it returns
the ast_channel_topic_all() topic instead of NULL. This can clean
up a lot of NULL checking we're doing currently.
* The fields Cause and Cause-txt were removed from the base channel
information and put only on the Hangup events, since those fields
are meaningless outside of a Hangup event.
* Removed the pipe-delimiter processing of the channelvars field,
since that's been deprecated forever.
(closes issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2405/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch started out simply as fixing the bouncing tests introduced
in r382685, but required some other changes to give it a decent
implementation.
To fix the bouncing tests, the UserEvent and Newexten AMI events
needed to be refactored to dispatch via Stasis. Dispatching directly
to AMI resulted in those events sometimes getting ahead of the
associated Newchannel events, which would understandably confuse anyone.
I found that instead of creating a zillion different message types and
structures associated with them, it would be preferable to define a
message type that has a channel snapshot and a blob of structured data
with a small bit of additional information. The JSON object model
provides a very nice way of representing structured data, so I went
with that.
* Move JSON support from res_json.c to main/json.c
* Made libjansson-dev a required dependency
* Added an ast_channel_blob message type, which has a channel
snapshot and JSON blob of data.
* Changed UserEvent and Newexten events so that they are dispatched
via ast_channel_blob messages on the channel's topic.
* Got rid of the ast_channel_varset message; used ast_channel_blob
instead.
* Extracted the manager functions converting Stasis channel events to
AMI events into manager_channel.c.
(issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2381/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change allows you to use XMPP buddy state in places where device state
can be used be used, such as dialplan hints. If at least one resource is
available the buddy is considered available. Now your phone can reflect
their IM status too!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added an option "discard_remote_hold_retrieval" (default "no") that if set does
not trigger the music on hold event. This essentially stops telling the peer
to start music on hold.
(issue ABE-2899)
Reported by: Denis Alberto Martinez
Review: https://reviewboard.asterisk.org/r/2336/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds support for RFC 3327 "Path" headers. This can be enabled in
sip.conf using the 'supportpath' setting, either on a global basis or on a
peer basis. This setting enables Asterisk to route outgoing out-of-dialog
requests via a set of proxies by using a pre-loaded route-set defined by the
Path headers in the REGISTER request. This patch also adds Realtime support
for dynamically updating the Path information for a peer.
A huge thank-you to Klaus Darillion and Olle E Johansson for their efforts
in writing this patch.
Review: https://reviewboard.asterisk.org/r/2235/
Review: https://reviewboard.asterisk.org/r/991/
(closes issue ASTERISK-16884)
Reported by: klaus3000
Tested by: klaus3000, oej, mjordan
patches:
path-1.8.0-patch.txt uploaded by klaus3000 (License 5054)
oolong-path-support-trunk in team branch by oej (License 5267)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The original report had to do with a realtime peer behind NAT being pruned and
the peer's private address being used instead of its external address. Upon
debugging, it was discovered that this was being caused by the addition of
the auto_force_rport and auto_comedia settings.
This patch does the following:
* Adds a missing note to the CHANGES file indicating that the default global nat
setting is auto_force_rport
* Constify the 'req' parameter for check_via()
* Add calls to check_via() in a couple of places in order for the auto_*
settings to do their job in attempting to determine if NAT is involved
* Set the flags SIP_NAT_FORCE_RPORT and SIP_PAGE2_SYMMETRICRTP if the auto_*
settings are in use where it was needed
* Moves the copying of peer flags up in build_peer() to before they are used;
this fixes the realtime prune issue
* Update the contrib/realtime schemas to allow the nat column to handle the
different nat setting combinations we have
This patch received a review and "Ship It!" on the issue itself.
(closes issue ASTERISK-20904)
Reported by: JoshE
Tested by: JoshE, Michael L. Young
Patches:
asterisk-20904-nat-auto-and-rt-peersv2.diff Michael L. Young (license 5026)
........
Merged revisions 382322 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For some channel drivers, specifically those that have a varying rate in the
number of audio samples, the audio quality for a MeetMe conference can be
exceedingly poor. This is due to a unilateral application of the DENOISE
function in func_speex to channels joining the conference.
The denoiser function in the speex library is initialized with the number of
audio samples in each sample that will be provided to it. If the number of
audio samples changes, the denoiser has to be thrown away and re-initialized.
While this could be worked around by removing func_speex, that doesn't help
if you actually use the denoiser with other channels on the system.
This patches does the following:
* Checks for the presence of func_speex as opposed to codec_speex when
determining if the DENOISE function is present (which is where the function
is actually implemented)
* Adds an option to MeetMe 'n' that causes the denoiser to not be applied
to a channel when it joins. This keeps the current behavior the default, but
let's users disable the denoiser if it causes problems on their system.
Review: https://reviewboard.asterisk.org/r/2358
(closes issue AST-1062)
Reported by: Thomas Arimont
........
Merged revisions 382227 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 382230 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382232 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, if one starts, stops, and then starts a recording again for a
conference the recorded data is appended to the file originally created
on the first record start. An option record_file_append has been added
that defaults to "yes", but when set to "no" will force creation of a new
file between every record start/stop.
(issue AST-1088)
Reported by: John Bigelow
Review: http://reviewboard.digium.internal/r/374/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These two variables were previously not being set when comebacktoorigin=yes
and the example configs seemed to imply that they should be. Since there
is no harm in this and since calls that are sent back to origin are capable
of continuing in the dialplan, this seemed like a no-brainer. Also it
supports some bridging tests I've been working on.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the capability for asynchronous manipulation of audio being
played back to a channel though a new AMI action "ControlPlayback". The
ControlPlayback action supports a number of operations, the availability of
which depend on the application being used to send audio to the channel.
When the audio playback was initiated using the ControlPlayback application
or CONTROL STREAM FILE AGI command, the audio can be paused, stopped,
restarted, reversed, or skipped forward. When initiated by other mechanisms
(such as the Playback application), the audio can be stopped, reversed, or
skipped forward.
Review: https://reviewboard.asterisk.org/r/2265/
(closes issue ASTERISK-20882)
Reported by: mjordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From the patch author:
"First this patch adds general support for busy detection. It also adds support
for the ECAM command at Sony Ericsson phones and also signals busy when only
early media was received but the call got not answered."
Review: https://reviewboard.asterisk.org/r/323
(closes issue ASTERISK-14527)
Reported by: Artem Makhutov
Tested by: Artem Makhutov
patches:
busy-full5.patch uploaded by artem (license 5757)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When ringinuse=no queue members can receive more than one call if these
calls happen at nearly the same time.
* Fix so a queue member does not receive more than one call from a queue.
NOTE: This fix does not prevent multiple calls to a member if the member
is in more than one queue.
* Did some refactoring to eliminate some code redundancy.
(issue ASTERISK-16115)
Reported by: nik600
Patches:
jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett
Modified
* Revert the -r341580 and -r341599 changes adding the queues.conf
check_state_unknown option as it was added in an attempt to fix this
problem. The fix did not need to be optional. The fix should not have
tried to explicitly set the device state. Setting the device state by
something other than the device introduces a race condition. I also could
not see how the change would be effective other than delaying the
app_queue code long enough for the device state to propagate to app_queue.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
BRIDGE_FEATURES did not previously support the automixmonitor feature. Now it
does. In addition, the BRIDGE_FEATURES variable would not apply features to
the proper party based on whether the feature option letter was in caps or
in lowercase (both ways would apply it to the caller). Now uppercase applies
to the caller while lowercase applies to the callee (like with the dial option)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new field is will show up within the response if the requested peer has a
subscribe context set.
(closes issue ASTERISK-20626)
Reported by: Jaco Kroon
Patches:
asterisk-sip-ami-SubscrContext.patch uploaded by jkroon (license 5671)
-with modifications by jrose to conform to style guidelines
Review: https://reviewboard.asterisk.org/r/2195/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
manager show commands now shows the full name of the command being displayed
regardless of size. The privilege column has also been removed from this
display. It will also now use the full length of the terminal if curses is
available. Manager show command will now always display the privilege of
the manager command within the CLI.
(closes ASTERISK-20396)
Reported by: Johan Wilfer
Review: https://reviewboard.asterisk.org/r/2143/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk's DTMF Specifications are based on AT&T specs, which may not be compatible in other countries.
Various countries have different specifications for the maximum power level differences
between the DTMF low group and high group of frequencies.
Power level difference between frequencies for different Administrations/RPOAs
NTT = Max. 5 dB
AT&T = 4dB(reverse) to 8dB(normal)
Danish = Max. 6 dB
Australian = Max. 10 dB
Brazilian = Max. 9 dB
ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1 (2006-03)
Now allow 4 variables to be individually configured in dsp.conf, with reasonable min/max of 2dB to 20dB.
Default is AT&T specifications
Add's the following variables to dsp.conf
;dtmf_normal_twist=6.31
;dtmf_reverse_twist=2.51
;relax_dtmf_normal_twist=6.31
;relax_dtmf_reverse_twist=3.98
(closes issue ASTERISK-20442)
Reported by: tbsky
Tested by: tbsky,alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2141/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds an optional header to the PlayDTMF AMI action, Duration.
It allows the duration of the DTMF digit to be played on the channel to be
specified in milliseconds.
(closes issue ASTERISK-18172)
Reported by: Renato dos Santos
patches:
send-dtmf.patch uploaded by Renato dos Santos (license #6267)
Modified slightly for this commit for Asterisk 12.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, if there are modifications to mailboxes that Asterisk is
not aware of, the user needs to add "pollmailboxes" to their mailbox
configuration, which repeatedly polls the subscribed mailboxes for
changes. This results in a lot of extra work for the CPU. This patch
introduces the AMI command VoicemailRefresh which permits external
applications to trigger the refresh themselves. The refresh can apply
to a specified mailbox only, an entire context, or all configured
mailboxes. Even a refresh performed on every mailbox would not consume
as much CPU as the pollmailboxes option, given that pollmailboxes runs
continuously and this only runs on demand.
(closes issue ASTERISK-17206)
(closes issue ASTERISK-19908)
Reported-by: Jeff Hutchins
Reported-by: Tilghman Lesher
Patch-by: Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This allows for the REDIRECTING dialplan function to be
used to set the reason to any string.
The SIP channel driver has been modified to set the redirecting
reason string to the value received in a Diversion header. In
addition, SIP 480 response reason text will set the redirecting
reason as well.
(closes issue AST-942)
reported by Malcolm Davenport
(closes issue AST-943)
reported by Malcolm Davenport
Review: https://reviewboard.asterisk.org/r/2101
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Sets INUSE when no free agents, NOT_INUSE when an agent is free.
modifes handle_statechange() scan members loop to scan for a free agent
and updates the Queue:queuename_avial devstate.
Previously exited early if the member was found in the queue.
Now Exits later when both a member was found, and a free agent was found.
alecdavis (license 585)
Reported by: Alec Davis
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/2121/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373188 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, if pause or unpause was issued on an interface
without specifying a specific queue, a PAUSEALL or UNPAUSEALL event
would be logged in the queue log even if that interface wasn't a
member of any queues. This patch changes it so that these events are
only logged when at least one member of any queue exists for that
interface.
(closes issue AST-946)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2079/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue ASTERISK-18390)
Reported by: Peter Racz
Patches:
dundi_cli_cache.patch.v2 uploaded by Peter Racz (license #6290)
ASTERISK-18390_dundi_cli_cache_jrose_mods_v2.diff uploaded by Jonathan Rose (license #6182)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is based on the work done by Olle Johansson on review board.
The idea is that the channel specified in an AMI originate or call
file is typically not connected to the outgoing extension until the
channel has been answered. With this change, an EarlyMedia header can
be specified for AMI originates and an early_media option can
be specified in call files. With this option set, once early media is
received on a channel, it will be connected with the outgoing extension.
(closes issue ASTERISK-18644)
Reported by Olle Johansson
Review: https://reviewboard.asterisk.org/r/1472
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Implementation of a dialplan function for checking manager accounts. Right now
it only returns the number of logged in sessions for a manager account, but
other attributes can be added later.
Patch by: Olle Johansson
Review: https://reviewboard.asterisk.org/r/421/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are
implemented in parallel to the existing numbered callgroup/pickupgroup
implementation. However, unlike the existing implementation, which is limited
to a maximum of 64 defined groups, the number of defined groups allowed for
named callgroups/pickupgroups is effectively unlimited.
Named groups are configured with the keywords "namedcallgroup" and
"namedpickupgroup". This corresponds to the numbered group definitions of
"callgroup" and "pickupgroup". Note that as the implementation of named groups
coexists with the existing numbered implementation, a defined named group of
"4" does not equate to numbered group 4.
Support for the named groups has been added to the SIP, DAHDI, and mISDN channel
drivers.
Review: https://reviewboard.asterisk.org/r/2043
Uploaded by:
Guenther Kelleter(license #6372)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is a patch from kkm from review board.
This is useful for adding headers to REFER requests that
emanate from a Transfer() dialplan application call.
This also fixes some uses of the Referred-by header, removing
an extra set of angle brackets.
I've modified the reporter's original patch to not require
any additions to the sip_refer header and to just remove the
referred_by_name from sip_refer since it is no longer needed
or used.
(closes Issue ASTERISK-17639)
reported by Kirill Katsnelson
Patches:
019059-sip-refer-addheaders-trunk-353549.diff
uploaded by Kirill Katsnelson (license #5845)
Review: https://reviewboard.asterisk.org/r/1159
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With this option set, channel variables can be set on
every manager originate. The Variable header can still
be used to set additional channel variables for individual
calls if desired.
This work was completed by Olle Johansson on review board.
I have applied the review feedback and am committing it in
order to get this into trunk before Asterisk 11 is branched.
Review: https://reviewboard.asterisk.org/r/1412
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From corruptor's review board posting:
"I've noticed that we can remove particular extension from context with
dialplan remove extension command but in order to remove all extensions
in the context we should delete them on by one. I've created dialplan
remove context command which uses ast_context_destroy to destroy the
whole context with all extensions. I've created to functions for in
pbx_config.c: handle_cli_dialplan_remove_context which actually removes
context and complete_dialplan_remove_context which completes input.
They are based on other similar functions and pretty trivial but I can be
mistaken somewhere.
"I've also modified dialplan add include <context2> into <context1>. I've
made it similar dialplan add extension ... command. It creates <context1>
if it doesn't exist and I've also modified complete_dialplan_add_include
and removed check for existance of <context2> because we can include
non-existent context into another one. (I usually include empty
(non-existent) contexts in advance). Should we raise warning in this case
as it's raised while reading extensions.conf?
"I use those functions with AMI. I think manager commands should be created
in addition to those CLI commands."
I've addressed the latest comments on review board and have made some other
coding guidelines-related cleanup. I also have modified the CHANGES file to
mention these new commands.
(closes issue ASTERISK-19292)
reported by Andrey Solovyev
Patches:
dialplan_add_include.patch
uploaded by Andrey Solovyev (license #5214)
dialplan_remove_context.patch
uploaded by Andrey Solovyev (license #5214)
Review: https://reviewboard.asterisk.org/r/2042
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch allows you to specify a port number for the MySQL server.
It's useful if a MySQL server is running on a non-standard port.
Even though this module is deprecated in favor of func_odbc, someone
asked for this feature and it seems pretty harmless to add.
It has been tested using a number of combinations of with/without a
port number specified in the dialplan and changing the port number
for mysqld.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch was submitted by mnicholson a while back. It adds a new AMI action
which allows users to request SIP peer status on demand similar to existing
PeerStatus events and to the output you would see from CLI with sip show peer
Review: https://reviewboard.asterisk.org/r/1098/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370518 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch changes the Asterisk configure script and build system to detect
the presence of the NetBSD editline library (libedit) on the system. If it is
found, it will be used in preference to the version included in the Asterisk
source tree.
(closes issue ASTERISK-18725)
Reported by: Jeffrey C. Ollie
Review: https://reviewboard.asterisk.org/r/1528/
Patches:
0001-Allow-linking-building-against-an-external-editline.patch uploaded by jcollie (license #5373) (heavily modified by kpfleming)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This updates the CHANGES file with things that were committed for
Asterisk 11, but were not noted in that file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370353 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds Named ACL functionality to Asterisk. This allows system
administrators to define an ACL and refer to it by a unique name. Configurable
items can then refer to that name when specifying access control lists.
It also includes updates to all core supported consumers of ACLs. That includes
manager, chan_sip, and chan_iax2. This feature is based on the deluxepine-trunk
by Olle E. Johansson and provides a subset of the Named ACL functionality
implemented in that branch. For more information on this feature, see acl.conf
and/or the Asterisk wiki.
Review: https://reviewboard.asterisk.org/r/1978/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This channel driver is a replacement for both chan_gtalk and chan_jingle but adds additional features not found in either.
These features include full configuration reload, video, full codec support, bidirectional cause code mapping, hold,
unhold, and ringing indication. It is also compliant with the current published Jingle and Google Jingle specifications.
The original Google Talk protocol is also supported for Google Voice interoperability.
You may ask yourself though where the name motif comes from... and I would say to you... music!
motif: a perceivable or salient recurring fragment or succession of notes
Sorta like a jingle!
Review: https://reviewboard.asterisk.org/r/1917/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new CLI command, 'stun show status'. This command will show
a table describing all known STUN servers and statuses.
(closes issue ASTERISK-18046)
Reported by: Jeremy Kister
Tested by: Jeremy Kister
patches:
(stun-show-status-v4-trunk.patch license #6232 uploaded by Jeremy Kister)
Review: https://reviewboard.asterisk.org/r/2001
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
options are documented in config sample
sample config rename to proper name - ooh323.conf
To change media address ooh323 send empty TCS if there was
completed TCS exchange or send facility forwardedelements
with new fast start proposal if not.
Then close transmit logical channels and renew TCS exchange.
If new fast start proposal is received then ooh323 stack call back
channel driver routine to change rtp address in the rtp instance.
If empty TCS is received then close transmit logical channels and
renew TCS exchange
Review: https://reviewboard.asterisk.org/r/1607/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Hangup handlers are an alternative to the h extension. They can be used
in addition to the h extension. The idea is to attach a Gosub routine to
a channel that will execute when the call hangs up. Whereas which h
extension gets executed depends on the location of dialplan execution when
the call hangs up, hangup handlers are attached to the call channel. You
can attach multiple handlers that will execute in the order of most
recently added first.
(closes issue ASTERISK-19549)
Reported by: Mark Murawski
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2002/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the core changes necessary to support AMI event documentation
in the source files of Asterisk, and adds documentation to those AMI events
defined in the core application modules. Event documentation is built from
the source by two new python scripts, located in build_tools:
get_documentation.py and post_process_documentation.py.
The get_documentation.py script mirrors the actions of the existing AWK
get_documentation scripts, except that it will scan the entirety of a source
file for Asterisk documentation. Upon encountering it, if the documentation
happens to be an AMI event, it will attempt to extract information about the
event directly from the manager event macro calls that raise the event. The
post_process_documentation.py script combines manager event instances that
are the same event but documented in multiple source files. It generates
the final core-[lang].xml file.
As this process can take longer to complete than a typical 'make all', it
is only performed if a new make target, 'full', is chosen.
Review: https://reviewboard.asterisk.org/r/1967/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds a number of methods for controlling the setting of 'ringinuse'
which is basically the same concept as the old ignorebusy setting,
only now the per member setting always controls whether or not the
member is actually ringed while in use. A CLI command and a manager
action have been added to change a given queue member's ringinuse
option while Asterisk is running and the an argument has been added
for adding members with deliberately set ringinuse in queues.conf
Some effort has been made to ensure compatability with dialplans and
databases still referring to 'ignorebusy'.
(issue ASTERISK-19536)
reported by: Philippe Lindheimer
Review: https://reviewboard.asterisk.org/r/1919/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch improves the handling of call id logging significantly with regard
to transfers and adding APIs to better handle specific aspects of logging.
Also, changes have been made to chan_sip in order to better handle the creation
of callids and to enable the monitor thread to bind itself to a particular
call id when a dialog is determined to be related to a callid. It then unbinds
itself before returning to normal monitoring.
review: https://reviewboard.asterisk.org/r/1886/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If enabled using the keepalive option in sip.conf a small packet will be sent
at a regular interval to keep the NAT mapping open. This is lightweight as the
remote side does not need to parse and handle a SIP message.
(closes issue AST-783)
Review: https://reviewboard.asterisk.org/r/1756/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add two new dialplan functions: FEATURE() and FEATUREMAP(). FEATURE()
lets you set some of the configuration options from the [general] section
of features.conf on a per-channel basis. FEATUREMAP() lets you customize
the key sequence used to activate built-in features, such as blindxfer,
and automon. See the built-in documentation for details.
Review: https://reviewboard.asterisk.org/r/1871/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk has a setting for the minimum allowed DTMF. If we get shorter
DTMF tones, these will be changed to the minimum on the outbound call
leg.
(closes issue ASTERISK-19772)
Review: https://reviewboard.asterisk.org/r/1882/
Reported by: oej
Tested by: oej
Patches by: oej
Thanks to the reviewers.
1.8 branch for this patch: agave-dtmf-duration-asterisk-conf-1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ISDN ETSI PTP and Q.SIG (And SS7 in future) have support for reporting who
was the original redirecting party of a call.
* Added support for the original redirecting party and reason to the
REDIRECTING function and the system core as well as to the stubbed
locations in sig_pri.c.
Review: https://reviewboard.asterisk.org/r/1829/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The current Security Events Framework API only supports IPv4 when it comes to
generating security events. This patch does the following:
* Changes the Security Events Framework API to support IPV6 and updates
the components that use this API.
* Eliminates an error message that was being generated since the current
implementation was treating an IPv6 socket address as if it was IPv4.
* Some copyright dates were updated on files touched by this patch.
(closes issue ASTERISK-19447)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
security_events_ipv6v3.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/1777/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Hangup now can take a regular expression as the Channel option. If you want
to hangup multiple channels, use /regex/ as the Channel option. Existing
behavior to hanging up a single channel is unchanged, but if you pass a regex,
the manager will send you a list of channels back that were hung up.
(closes issue ASTERISK-19575)
Reported by: Mark Murawski
Tested by: Mark Murawski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added 'b' and 'B' options to Dial. These options will allow you to run
last-minute dialplan on the caller and callee channels while the Dial
application is executing, but before the call is started. For example you
can use the 'b' option to run dialplan on the callee channel to get the name
of the newly created channel right away.
Review: https://reviewboard.asterisk.org/r/1229/
(closes issue: ASTERISK-19548)
Reported by: Mark Murawski
Tested by: Mark Murawski, Stefan Schmidt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For analog lines, enables Asterisk to use dialtone detection per channel
if an incoming call was hung up before it was answered. If dialtone is
detected, the call is hung up.
no: Disabled. (Default)
yes: Look for dialtone for 10000 ms after answer.
<number>: Look for dialtone for the specified number of ms after answer.
always: Look for dialtone for the entire call. Dialtone may return
if the far end hangs up first.
dialtone_detect=yes
dialtone_detect=5000
dialtone_detect=always
(closes issue ASTERISK-19316)
Reported by: Jeremy Pepper
Patch by: Jeremy Pepper
Tested by: rmudgett,Jeremy Pepper
Review: https://reviewboard.asterisk.org/r/1737/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
send_diversion=no will prevent Diversion headers from being added to SIP
requests. This doesn't prevent Diversion from being added with dialplan
such as with SIPAddHeader.
(closes issue ASTERISK-16862)
Reported by: rsw686
Review: https://reviewboard.asterisk.org/r/1769/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change permits each verbose destination (consoles, logger) to have its
own concept of what the verbosity level is. The big feature here is that
the logger will now be able to capture a particular verbosity level without
condemning each console to need to suffer that level of verbosity.
Additionally, a stray 'core set verbose' will no longer change what will go
to the log.
Review: https://reviewboard.asterisk.org/r/1599/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a variable AGIEXITONHANGUP for res_agi. If this variable is
set to "yes" on a channel, AGI() will exit immediately once a channel hangup
has been detected. This was the behavior of AGI() in Asterisk 1.4 and earlier
and is still desired by some people.
Review: https://reviewboard.asterisk.org/r/1734/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the auto_force_rport and auto_comedia NAT options. It
also converts the nat= setting to a list of comma-separated combinable
options: no, force_rport, comedia, auto_force_rport, and auto_comedia.
nat=yes remains as an undocumented option equal to
"force_rport,comedia". The first instance of 'yes' or 'no' in the list
stops parsing and overrides any previously set options. If an auto_*
option is specified with its non-auto_ counterpart, the auto setting
takes precedence.
This patch builds upon the patch posted to ASTERISK-17860 by JIRA user
pedro-garcia.
(closes issue ASTERISK-17860)
Review: https://reviewboard.asterisk.org/r/1698/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is based on the one by David Vossel, developer extrodinaire, at
https://reviewboard.asterisk.org/r/344/. If multiple peers are defined with the
same host/port, but differing callbackextensions, it chooses the peer with the
matching callbackextension. Since callbackextension creates an outbound
registration with the callbackextension as the Contact address, matching an
incoming request by that (in addition to the host/port) makes a lot of sense.
This patch also adds support for callbackextension to realtime by querying all
peers with callbackextensions on reload and adding registrations for them.
(closes issue ASTERISK-13456)
Review: https://reviewboard.asterisk.org/r/344/
Review: https://reviewboard.asterisk.org/r/1717/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch removes res_ais and introduces a new module, res_corosync.
The OpenAIS project is deprecated and is now just a wrapper around
Corosync. This module provides the same functionality using the same
core infrastructure, but without the use of the deprecated components.
Technically res_ais could have been used with an AIS implementation other
than OpenAIS, but that is the only one I know of that was ever used.
Review: https://reviewboard.asterisk.org/r/1700/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When Asterisk is used with various third-party libraries (CURL, PostgresSQL,
many others) that have the ability themselves to use OpenSSL, it is possible
for conflicts to arise in how the OpenSSL libraries are initialized and
shutdown. This patch addresses these conflicts by 'wrapping' the important
functions from the OpenSSL libraries in a new shared library that is part
of Asterisk itself, and is loaded in such a way as to ensure that *all*
calls to these functions will be dispatched through the Asterisk wrapper
functions, not the native functions.
This new library is optional, but enabled by default. See the CHANGES file
for documentation on how to disable it.
Along the way, this patch also makes a few other minor changes:
* Changes MODULES_DIR to ASTMODDIR throughout the build system, in order to
more closely match what is used during run-time configuration.
* Corrects some errors in the configure script where AC_CHECK_TOOLS was used
instead of AC_PATH_PROG.
* Adds a new variable for linker flags in the build system (DYLINK), used for
producing true shared libraries (as opposed to the dynamically loadable
modules that the build system produces for 'regular' Asterisk modules).
* Moves the Makefile bits that handle installation and uninstallation of the
main Asterisk binary into main/Makefile from the top-level Makefile.
* Moves a couple of useful preprocessor macros from optional_api.h to
asterisk.h.
Review: https://reviewboard.asterisk.org/r/1006/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA
function in the authorization attempt.
* Pass up better From header contents for SIP to use. Now is in the
"display-name" <URI> format expected by MessageSend. (Note that this is a
behavior change that could concievably affect some people.)
* Block user from adding standard headers that are added automatically.
(To, From,...)
* Allow the user to override the Content-Type header contents sent by
MessageSend.
* Decrement Max-Forwards header if the user transferred it from an
incoming message.
* Expand SIP short header names so the dialplan and other code only has to
deal with the full names.
* Documents what SIP expects in the MessageSend(from) parameter.
(closes issue ASTERISK-18992)
Reported by: Yuri
(closes issue ASTERISK-18917)
Reported by: Shaun Clark
Review: https://reviewboard.asterisk.org/r/1683/
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This is a feature patch which allows an 'announcement' option to be specified in
musiconhold.conf which should be set to the name of a sound. If a valid sound is
specified for this option, then it will be played on that music on hold class whenever
a channel bound to that class is put on hold as well as when Asterisk is able to detect
that a song has ended before starting the next song (excludes external players).
(closes ASTERISK-18977)
Reported by: Timo Teräs
Patches:
asterisk-moh-announcement.diff uploaded by Timo Teräs (license 5409)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352134 65c4cc65-6c06-0410-ace0-fbb531ad65f3
MixMonitor receives a new option i(channel_variable) which stores the unique id at said
variable. StopMixMonitor now accepts ID as an optional argument, which if included will
make StopMixMonitor specifically target the mixmonitor on that particular channel. CLI
commands and AMI actions have been ammended to work with the IDs as well. In addition,
monitors across a channel can now be listed be listed via CLI command "mixmonitor list
<channel>" which will display all of the mixmonitors active on that channel along with
the files they each have open. Created by Sergio González Martín.
(closes issue ASTERISK-19096)
Reported by: Sergio González Martín
Review: https://reviewboard.asterisk.org/r/1643/
Review: https://reviewboard.asterisk.org/r/1682/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Adds per-parking lot options comebackcontext and comebackdialtime
* Makes comebacktoorigin settable per parking lot
* Sets a PARKER channel variable when comebacktoorigin is disabled
(closes issue ASTERISK-16643)
Reported by: Mitch Sharp (bluecrow76)
Patches:
asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff by Mitch Sharp (bluecrow76) license 5231
with updates by me.
Review: https://reviewboard.asterisk.org/r/1674
Review: https://reviewboard.asterisk.org/r/963
Reviewed by Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In order to better handle RTP sources with strictrtp enabled (which is now default in 10)
using the learning mode to figure out new sources when they change is handled by checking
for a number of consecutive (by sequence number) packets received to an rtp struct
based on a new configurable value called 'probation'. Also, during learning mode instead
of liberally accepting all packets received, we now reject packets until a clear source
has been determined.
Review: https://reviewboard.asterisk.org/r/1663/
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This patch allows the imapserver, imapport, and imapflags settings to be
overridden for any voicemail user. It also documents the settings in
the sample voicemail.conf file, and updates the voicemail schema to
allow storage of those columns.
(closes issue ASTERISK-16489)
Reporter: Hubert Mickael
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1614/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch moves destruction of sip peers to immediately after the general section of
sip.conf is read so that autocreatepeer setting can be read before deletion of peers.
If autocreatepeer=persist at reload, then peers created by the autocreatepeer setting
will be skipped when purging the current SIP peer list.
(closes ASTERISK-16508)
Reported by: Kirill Katsnelson
Patches:
017797-kkm-persist-autopeers-1.8.patch uploaded by Kirill Katsnelson (license 5845)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In order to check the availability of the caller's name, app_voicemail will check for an
audio file in <astspooldir>/recordings/callerids/
This change sets a precedent for where to put recordings of names. Currently the idea is
that recordings here could also be used for applications like confbridge and meetme to
find recorded names in this folder from callerid (when another recording isn't available)
(closes issue ASTERISK-18565)
Reporter: Russell Brown
Patches:
r uploaded by Russel Brown (license 6182)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
During testing, it was discovered that there were a number of side effects
introduced by r346391 and subsequent check-ins related to it (r346429,
r346617, and r346655). This included the /main/stdtime/ test 'hanging',
as well as the remote console option failing to receive the appropriate output
after a period of time.
I only backed out the changes to main/ and utils/, as this was adequate
to reverse the behavior experienced.
(issue ASTERISK-18974)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a caller sends DTMF while the SayUnixTime application is saying the time, The call
would jump to the next extension much like it does during Background(). This patch adds
option 'j' to SayUnixTime which when used employs the old behavior. Also, this patch
allows arguments to sayunixtime to not be used as empty strings in the case of something
like 'sayunixtime(,,,j)' or 'sayunixtime(,,pattern).
(closes issue ASTERISK-16675)
Reported by: jlpedrosa
Patches:
patch_SayUnixTime_noJump.patch uploaded by jlpedrosa (license 5959)
Review: https://reviewboard.asterisk.org/r/956/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds two new menu features to app_confbridge, admin_toggle_menu_
participants and participant_count. The admin action will globally mute /
unmute all non-admin participants on a converence, while the participant
count simply exposes the existing participant count function to the
conference bridge menu.
This also adds configuration options to change the sound played when the
conference is globally muted / unmuted, as well as the necessary config
hooks to place these functions in the DTMF menus.
(closes issue ASTERISK-18204)
Reported by: Kevin Reeves
Tested by: Matt Jordan
Patches:
app_confbridge.c.patch.txt, conf_config_parser.c.patch.txt,
confbridge.h.patch.txt uploaded by Kevin Reeves (license 6281)
Review: https://reviewboard.asterisk.org/r/1518/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
You can now use the built in variables , , and
within a dynamic weight. For example, this could be useful when you want
to pass requested lookup number to the SHELL() function which could be
used to execute a script to dynamically set the weight of the result.
(Closes issue ASTERISK-13657)
Reported by: Joel Vandal
Tested by: Leif Madsen, Russell Bryant
Patches:
asterisk-1.6-dundi-varhead.patch uploaded by Joel Vandal (License #5374)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343693 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
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r341580 | irroot | 2011-10-20 19:13:23 +0200 (Thu, 20 Oct 2011) | 15 lines
Add option to check state when state is unknown
r341486 reverts r325483 this is a rework of the patch.
optimize to minimize load.
add option check_state_unknown to control whether a member with unknown
device state is checked there is a small % chance that calls will be sent
to the member when they on a call.
app_queue will see a device with unknown state as available and does not
try verify the state without this option enabled.
Review: https://reviewboard.asterisk.org/r/1535/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added func FAXOPT(faxdetect)=yes,cng,t38[,timeout]/no
to enable dialplan faxdetect allowing more flexibility.
as soon as a fax tone is detected the framehook is removed.
there is a penalty involved in running this framehook on
non G711 channels as they will be transcoded.
CNG tone is suppresed using the SQUELCH flag to allow
WaitForNoise to be run on the channel to detect Voice.
(Closes issue ASTERISK-18569)
Reported by: Myself
Reviewed by: Matthew Nicholson, Kevin Fleming
Review: https://reviewboard.asterisk.org/r/1116/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339465 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
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r337595 | jrose | 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) | 12 lines
Generate Security events in chan_sip using new Security Events Framework
Security Events Framework was added in 1.8 and support was added for AMI to generate
events at that time. This patch adds support for chan_sip to generate security events.
(closes issue ASTERISK-18264)
Reported by: Michael L. Young
Patches:
security_events_chan_sip_v4.patch (license #5026) by Michael L. Young
Review: https://reviewboard.asterisk.org/r/1362/
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r337597 | jrose | 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) | 10 lines
Forgot to svn add new files to r337595
Part of Generating security events for chan_sip
(issue ASTERISK-18264)
Reported by: Michael L. Young
Patches:
security_events_chan_sip_v4.patch (License #5026) by Michael L. Young
Reviewboard: https://reviewboard.asterisk.org/r/1362/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
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r337219 | oej | 2011-09-21 11:32:50 +0200 (Ons, 21 Sep 2011) | 13 lines
Make ast_pbx_run() not default to s@default if extension is not found
Review: https://reviewboard.asterisk.org/r/1446/
This is a bug - or architecture mistake - that has been in Asterisk for a
very long time. It was exposed by the AMI originate action and possibly
some other applications. Most channel drivers checks if an extension
exists BEFORE starting a pbx on an inbound call, so most calls will
not depend on this issue.
Thanks everyone involved in the review and on IRC and the mailing list
for a quick review and all the feedback.
(closes issue ASTERISK-18578)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
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r337178 | oej | 2011-09-21 10:51:41 +0200 (Ons, 21 Sep 2011) | 14 lines
Change strictrtp option to default to yes in the RTP module
Suggested by Kapejod on Facebook
Review: https://reviewboard.asterisk.org/r/1448/
(closes issue ASTERISK-18587)
Thanks for quick feedback to kpfleming and Tilghman
--Denna och nedanstående rader kommer inte med i loggmeddelandet--
M CHANGES
M configs/rtp.conf.sample
M res/res_rtp_asterisk.c
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337179 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) | 14 lines
Allow Setting Auth Tag Bit length Based on invite or config option
Update the SIP SRTP API to allow use of 32 or 80 bit taglen.
Curently only 80 bit is supported.
The outgoing invite will use the taglen of the incoming invite preventing
one-way audio.
(Closes issue ASTERISK-17895)
Review: https://reviewboard.asterisk.org/r/1173/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336937 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
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r336042 | oej | 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) | 12 lines
Meetme: Introducing a new option "k" to kill a conference if there's only a single member left.
When using Meetme as a modular call bridge from third party applications, it's handy to make
it behave like a normal call bridge. When the second to last person exists, the last person
will be kicked out of the conference when this option is enabled.
(closes issue ASTERISK-18234)
Review: https://reviewboard.asterisk.org/r/1376/
Patch by oej, sponsored by ClearIT, Solna, Sweden
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This permits the list of codecs to be specified in one configuration line,
instead of two or more, generally with the aim of either allowing all codecs
with the exception of a few or disallowing most but permitting a few.
Review: https://reviewboard.asterisk.org/r/1411/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Matthew Jordan
Mark Michelson
George Joseph
Joshua Colp
Richard Mudgett
Jason Parker
Jonathan Rose
Michael L. Young
Kinsey Moore
Scott Griepentrog
Igor Goncharovskiy
Russell Bryant
Kevin Harwell
Walter Doekes
David M. Lee
Olle Johansson
Alec L Davis
Terry Wilson
Kevin P. Fleming
Alexandr Anikin
Sean Bright
Mark Murawki
Tilghman Lesher
Paul Belanger
Leif Madsen
Gregory Nietsky