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This commit introduces official support for R2 signaling in chan_dahdi. The
modifications to chan_dahdi, and the supporting library, LibOpenR2, were both
written by Moises Silva.
Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6
in Brazil, México and Argentina. An unknown number of users (but at least 1)
are using it in each of the following countries: Colombia, Nepal, Thailand,
Venezuela, Perú, and probably others.
To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/.
Information about configuration can be found in configs/chan_dahdi.conf.sample.
The code committed is the most up to date version, which was being maintained
in svn/asterisk/team/moy/mfcr2/.
I would also like to include a Thank You to the many others that tested this
code beyond those listed in this commit message. These are the names that I
could find in the mantis issue.
(closes issue #12509)
Reported by: moy
Patches:
chan_zap-mfr2.patch uploaded by moy (license 222)
Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen
Review: http://reviewboard.digium.com/r/40/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182355 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #14646)
Reported by: strk
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181499 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar 2009) | 25 lines
Fix broken mailbox parsing when searchcontexts option is enabled.
When using the searchcontexts option in voicemail.conf, the code
made the assumption that all mailbox names defined were unique across
all contexts. However, the code did nothing to actually enforce this
assumption, nor did it do anything to alert a user that he may have
created an ambiguity in his voicemail.conf file by defining the same
mailbox name in multiple contexts.
With this change, we now will issue a nice long warning if searchcontexts
is on and we encounter the same mailbox name in multiple contexts and ignore
any duplicates after the first box. Whether searchcontexts is enabled or not,
if we come across a duplicate mailbox in the same context, then we will issue
a warning and ignore the duplicated mailbox. I have also added a small note
to voicemail.conf.sample in the explanation for searchcontexts explaining
that you cannot define the same mailbox in multiple contexts if you have
enabled the option.
(closes issue #14599)
Reported by: lmadsen
Patches:
14599.patch uploaded by mmichelson (license 60) (with slight modification)
Tested by: lmadsen
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When the subscription context for a call pickup subscription differs
from the context of the call pickup target, there's not an easy way
to divine what context should be used for the pickup. The way to work
around this is to use PICKUPMARK as the context for the pickup.
This has been documented in the sip.conf.sample file
(ABE-1708)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines
Clarify some documentation of queues.conf.sample
It had always been possible to explicitly specify a "blank"
value for a sound file in queues.conf and have no sound played
back. The problem with this is that it would result in some ugly
CLI warnings from file.c.
This commit introduces a check when playing a file in app_queue
to see if the name of the file is zero-length and return early if
that is the case. Also, the ability to specify the blank sound
files in queues.conf is now mentioned more clearly in queues.conf.sample
(closes issue #14227)
Reported by: caspy
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https://origsvn.digium.com/svn/asterisk/branches/1.4
In this case, it's just a matter of reducing the default timeouts from 2000
to 1000 msec, as the max def feature digit timeout is no longer halved.
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r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines
This change moves the default feature digit timeout to 1000 ms from the previous default of 500.
As per bug 14515, a dev discussion arrived at a "mediated concensus"
of a default feature digit timeout of 1.0 sec. Some voted for 1300;
ctooley thought 1500 for distracted phone users in phone booths;
kpfleming put his foot down at 1.0 sec.
Users who found the previous default max delay of 250 msec perfect,
are welcome to override the new default. Notice that I said that
250 msec was the default; wait a minute, you might say, the config
file said it was 500 msec!; well, because of the bug fix for 14515,
we found that 500 msec was actually enforcing a max of 250. The bug
fix would restore 500 msec, but we felt even that was a bit tight
for most users... 2000 msec was pushed earlier by mmichelson, so
that reduces to 1000 msec after the bug fix. Enjoy!
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@178986 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #13826)
Reported by: azielke
Patches:
pickupsound2-trunk.patch uploaded by azielke (license 548)
__20081124_bug_13826_updated.patch uploaded by lmadsen (license 10)
Tested by: lmadsen
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009) | 5 lines
Add section about the #exec command in configuration files.
(closes issue #14540)
Reported by: jtodd
Patch by: jtodd, with additional notes by tilghman (license 14)
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(closes issue #14372)
Reported by: fhackenberger
Patches:
voicemail_individual_subject_and_body_1.6.1 uploaded by fhackenberger (license 592)
with additional fixes by Corydon76 (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@178107 f38db490-d61c-443f-a65b-d21fe96a405b
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This patch includes a number of changes to the indications API. The primary
motivation for this work was to improve stability. The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.
The changes included are:
1) Remove the module res_indications. This included the critical functionality
that actually loaded the indications configuration. I have seen many people
have Asterisk problems because they accidentally did not have an
indications.conf present and loaded. Now, this code is in the core,
and Asterisk will fail to start without indications configuration.
There was one part of res_indications, the dialplan applications, which did
belong in a module, and have been moved to a new module, app_playtones.
2) Object management has been significantly changed. Tone zones are now
managed using astobj2, and it is no longer possible to crash Asterisk by
issuing a reload that destroys tone zones while they are in use.
3) The API documentation has been filled out.
4) The API has been updated to follow our naming conventions.
5) Various bits of code throughout the tree have been updated to account
for the API update.
6) Configuration parsing has been mostly re-written.
7) "Code cleanup"
The code is from svn/asterisk/team/russell/indications/.
Review: http://reviewboard.digium.com/r/149/
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Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed.
Review: http://reviewboard.digium.com/r/159/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175597 f38db490-d61c-443f-a65b-d21fe96a405b
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When the 'faxdetect' configuration option is used, one may also want to use
the 'faxbuffers' configuration option in chan_dahdi.conf. This option will
dynamically use the configured 'faxbuffers' buffer policy on a channel for
the life of the call following the detection of fax tones. The faxbuffers
buffer policy will be reverted during call teardown.
An example use of 'faxbuffers' is below. This example would switch to using
6 buffers with a full buffer policy.
faxbuffers=>6,full
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175411 f38db490-d61c-443f-a65b-d21fe96a405b
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This patch adds forceencryption=yes as an iax.conf option. When force encryption is enabled, no unencrypted connections are allowed. This insures all connections are encrypted. This is a new feature, so CHANGES and iax.conf.sample are updated as well.
(closes issue #13285)
Reported by: sgofferj
Tested by: russell
Review: http://reviewboard.digium.com/r/150/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175344 f38db490-d61c-443f-a65b-d21fe96a405b
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This is very similar to the DAHDI immediate=yes option. When the phone is picked up, instead of giving a dialtone it connects directly to the "s" extension. Changes where implemented in chan_iax2.c to directly connect to the "s" extension in the appropriate context when this option is enabled. Examples explaining its use are added to iax2.conf.sample. CHANGES has been updated as well.
(closes issue #14266)
Reported by: jcovert
Patches:
chan_iax2.c.patch-trunk uploaded by jcovert (license 551)
iax.conf.sample.patch uploaded by jcovert (license 551)
Tested by: jcovert, dvossel
Review: http://reviewboard.digium.com/r/143/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@174046 f38db490-d61c-443f-a65b-d21fe96a405b
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In trunk, the only necessary change pointed out was that the call
to ChanIsAvail uses an option that has been removed.
For the 1.6.1 branch, however, it appears that the sample file is
badly in need of updating since there are |'s used all over the place
there. My tentative plan is just to copy trunk's sample config file
to those branches since the info there is most up-to-date and should
be correct for use in 1.6.1
Thanks to macli in #asterisk-dev for bringing this up
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173776 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009) | 5 lines
Add warning to standard config, that globals may be overridden by other
dialplan configuration files.
(closes issue #14388)
Reported by: macli
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173104 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #13861)
Reported by: scramatte
Patches:
__20080110-res_ldap.conf-2.patch uploaded by blitzrage (license 10)
Tested by: jcovert
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172894 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
Fix feature inheritance with builtin features
When using builtin features like parking and transfers, the AST_FEATURE_* flags
would not be set correctly for all instances when either performing a builtin
attended transfer, or parking a call and getting the timeout callback. Also,
there was no way on a per-call basis to specify what features someone should
have on picking up a parked call (since that doesn't involve the Dial() command).
There was a global option for setting whether or not all users who pickup a
parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
AUTOMON, or PARKCALL.
This patch:
1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
dialplan or with setvar in channels that support it. This variable can be set
to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
equivalent dial options), to set what features should be activated on this
channel. The patch moves the setting of the features datastores into the
bridging code instead of app_dial to help facilitate this.
2) adds global options parkedcallparking, parkedcallhangup, and
parkedcallrecording to be similar to the parkedcalltransfers option for
globally setting features.
3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
extension since tracking everything through multiple masquerades, etc. is
difficult and error-prone
4) attempts to fix all cases of return calls from parking and completed builtin
transfers not having the correct permissions
(closes issue #14274)
Reported by: aragon
Patches:
fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
Tested by: aragon, otherwiseguy
Review http://reviewboard.digium.com/r/138/
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* Added doxygen comments to the major dahdi structures.
* Fixed PRI and SS7 using an incorrect string value if the extension
delimiter is not present in the Dial() function.
* Fixed SS7 not checking if the dialed extension is at least as long
as the stripmsd option.
* Fixed PRI not handling unknown TON/NPI prefix letters correctly.
* Fixed some uninitialized string variables on FXS ports.
configs/chan_dahdi.conf.sample
* Updated some documentation.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172400 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines
Add a better explanation of the difference between the device namespace and the dialplan for newbies.
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(closes issue #14325)
Reported by: DEA
Patches:
skinny.conf.sample-trunk.txt uploaded by DEA (license 3)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@171043 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24 Jan 2009) | 2 lines
Remove superfluous implementation note (closes issue #14319)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan 2009) | 8 lines
Add notes to the idlecheck explanation in res_odbc.conf.sample
(closes issue #14319)
Reported by: klaus3000
Patches:
patch_idlecheck_res_odbc.conf.sample.txt uploaded by klaus3000 (license 65)
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ability.
(closes issue #14104)
Reported by: alecdavis
Patches:
asttrunk-14104.diff2.txt uploaded by dbailey (license )
chan_dahdi.rpas_and_fsk.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, dbailey
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@169277 f38db490-d61c-443f-a65b-d21fe96a405b
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This prevents the situation when MWI messages are added to caller ID spills causing the channel to be hung up
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@169153 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168721 | oej | 2009-01-15 19:43:43 +0100 (Tor, 15 Jan 2009) | 2 lines
Meetme actually has realtime but wasn't documented
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refer to a peer
in the register= line.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168711 f38db490-d61c-443f-a65b-d21fe96a405b
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sip.conf
With this commit, a register => line in sip.conf may contain a port number in the
"user" section of the line. Please see CHANGES and sip.conf.sample for more
details regarding this.
(closes issue #14198)
Reported by: Nick_Lewis
Patches:
chan_sip.c-domainport2.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168575 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009) | 2 lines
s/ringdance/ringcadence/ for Bulgaria
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Update the queues.conf.sample documentation to mention that you need to preload chan_local.so as well if you plan on using Local channels for queue members, and you're preloading pbx_config.so.
(closes issue #14179)
Reported by: CrashHD
Tested by: CrashHD
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@167477 f38db490-d61c-443f-a65b-d21fe96a405b
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enabled (either globally or for a specific peer), chan_sip will treat any SDP
data it receives as new data and update the media stream accordingly. By
default, Asterisk will only modify the media stream if the SDP session version
received is different from the current SDP session version. This option is
required to interoperate with devices that have non-standard SDP session
version implementations (observed by toc on the bug tracker with Microsoft OCS
which always uses 0 as the session version).
http://reviewboard.digium.com/r/94/
(closes issue #13958)
Reported by: toc
Tested by: toc
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165180 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #14028)
Reported by: mostyn
Patches:
timezone-v2.patch uploaded by mostyn (license 398)
(with additional code guideline fixes and a memory leak fix by me - license 14)
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to poke all peers at seemingly the same time.
(closes issue #13217)
Reported by: cervajs
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164809 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #13698)
Reported by: fhackenberger
Patches:
20081211__bug13698.diff.txt uploaded by Corydon76 (license 14)
Tested by: fhackenberger
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forwarded as not urgent.
(closes issue #14063)
Reported by: jaroth
Patches:
urgfwd_v2.patch uploaded by jaroth (license 50)
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after T38 is negotiated.
Terry Wilson created the original patch for this functionality, which I slightly modified and added
the faxdetect=yes|no configuration option. This patch is only for T38 fax detection and does not
do anything for G711 over SIP fax detection. By default, this option is disabled.
Reviewboard: http://reviewboard.digium.com/r/69/
This functionality is for issue AST-140.
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Based on cli_permissions.conf configuration file, we are able to permit or deny
cli commands based on some patterns and the local user and group running rasterisk.
(Sorry if I missed some of the testers).
Reviewboard: http://reviewboard.digium.com/r/11/
(closes issue #11123)
Reported by: eliel
Tested by: eliel, IgorG, Laureano, otherwiseguy, mvanbaak
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until the originating line hangs up.
(closes issue #12382)
Reported by: one47
Patches:
zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license 23)
zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed (license 463)
Tested by: fleed
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@159317 f38db490-d61c-443f-a65b-d21fe96a405b
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is only found if it dialed the extension that was subscribed to. You can now
specify 'ignore-context' for the 'notifycid' option in sip.conf which will, as
it's value implies, ignore the current context of the caller when doing the
lookup.
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