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This is a continuation of revision 885 to LibPRI (Capture and expose the Reverse
Charging Indication IE on ISDN PRI) which added the ability to get/set Reverse
Charging Indication in LibPRI. This patch adds the ability to specify RCI on
the outbound leg of a PRI call from within Asterisk, by prefixing the dialed
number with a capital 'C' like:
...,Dial(DAHDI/g1/C4445556666)
And to read it off an inbound channel:
exten => s,1,Set(RCI=${CHANNEL(reversecharge)})
Thanks again to rmudgett for the thorough review.
(closes issue #13760)
Reported by: mrgabu
Review: https://reviewboard.asterisk.org/r/303/
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.sample).
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Someone asked yesterday, "is there a good reason why we can't just put these
modules in Asterisk?". After a brief discussion, as long as the modules are
clearly set aside in their own directory and not enabled by default, it is
perfectly fine.
For more information about why a module goes in addons, see README-addons.txt.
chan_ooh323 does not currently compile as it is behind some trunk API updates.
However, it will not build by default, so it should be okay for now.
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This might seem like a legitimate comment that merely needed semicolon
prefixes, but in reality, the adaptive layer is designed to allow arbitrary
CDR variables, without needing the use of a userfield to store multiple items.
It's therefore not only invalid syntax but also goes against the intent of the
adaptive method.
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The original patch for this was written by Brett Bryant, and I split it out into
it's own module.
(closes issue #12876)
Reported by: bbryant
Patches:
06162008_cdr_custom_syslog.diff uploaded by bbryant (license 36)
05212009_cdr_syslog.patch uploaded by seanbright (license 71)
Tested by: seanbright
Review: https://reviewboard.asterisk.org/r/297/
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configurable values for finer control.
(closes issue #8855)
Reported by: mikma
Tested by: klaus3000, file
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application and channel.
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CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
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GR-303.
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Also change the preferred configuration option from 'hostname' (which was
misleading because it didn't actually treat the value as a hostname) to
'connection' and added some verbage explaining that the user would need to
refer to their freetds.conf file for those settings. 'hostname' was kept
as a backwards compatible configuration parameter.
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chan_sip has an option to save the sysname on rtupdate. This patch copies that same logic to chan_iax.
(closes issue #14837)
Reported by: barthpbx
Patches:
iax2-rtsavesysname.patch uploaded by barthpbx (license 744)
rt_iax.diff uploaded by dvossel (license 671)
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the new skip category feature unless supported
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realize this was never done but was working anyways
also added support for skip category request feature of openr2 and updated chan_dahdi.conf.sample
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should be specified.
(closes issue #14367)
Reported by: Nick_Lewis
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This code was there because of the AgentCallbackLogin() application.
->loginchan[] member was only used by AgentCallbackLogin().
Agent where dumped to astdb if they where logged in using AgentCallbacklogin()
so they are not being dumper anymore.
Review: https://reviewboard.asterisk.org/r/267/
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This commit add Calendaring support to Asterisk for iCalendar, CalDAV, and MS
Exchange calendars. Exchange support has only been tested on Exchange Server 2k3
and does not support forms-based authentication at this time (patches *very*
welcome). Exchange support is also currently missing the ability to return a
list of a meting's attendees (again, patches are very, very welcome).
Features include:
Querying a calendar for events over a specific time range
Checking a calendar's busy status via the dialplan
Writing calendar events via the dialplan (CalDAV and Exchange only)
Handling calendar event notifications through the dialplan
(closes issue #14771)
Tested by: lmadsen, twilson, Shivaprakash
Review: https://reviewboard.asterisk.org/r/58
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Let's try that again, this time removing trailing whitespace and not leading
whitespace. I can't believe no one noticed.
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the sample configuration files.
(closes issue #15207)
Reported by: seandarcy
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In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset.
(closes issue #12282)
Reported by: rjain
Patches:
reg_patch_1.diff uploaded by dvossel (license 671)
Tested by: dvossel
(closes issue #14727)
Reported by: pj
Patches:
reg_patch_3.diff uploaded by dvossel (license 671)
Tested by: pj, dvossel
Review: https://reviewboard.asterisk.org/r/249/
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functionality (allowing multiple mappings).
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SIP purists may want to look the other way...
When COLP/CONP support for SIP was committed, there was a condition under
which Asterisk may transmit a SIP UPDATE in order to communicate the change
in connected line information. The issue here is that while we could send a
SIP UPDATE message, we were not prepared to receive such an UPDATE and would
always responde with a 501 when we received an UPDATE.
The situation was a bit rough. We really want to be able to receive UPDATEs
having to do with connected line changes, but the amount of effort involved
in properly supporting RFC 3311 was staggering. This commit represents a
compromise.
First, it was decided that it is important to only send a SIP UPDATE to
an endpoint that is able to handle one. So, now we have added parsing of
the Allow header into SIP. We store the allowed methods on SIP peers so
that when we communicate with them, we already will know what we can and
cannot send to them. We will parse the peer's allowed methods when he registers
with us. If the peer is not the type to register with us, but the qualify option
is enabled, then we will use the response to the OPTIONS request we send
the peer to determine the peer's allowed methods. When the peer's registration
expires, or when qualify deems the peer to be unreachable, we clear the allowed
methods from the peer.
For an actual call, we will copy the peer's allowed methods to the sip_pvt
representing the call leg. If we are communicating with an endpoint which is
not a peer, then we will just parse the Allow header from the first message
we receive during the call and store the information in the sip_pvt.
If, during communication with a peer, we receive a 501 response, then we will
make sure to save the fact that we cannot use that method when communicating
with that peer.
Now, with all that infrastructure in place, the only actual place we use this
information currently is when attempting to send a connected line change using
an UPDATE request. If we cannot send the change immediately using an UPDATE,
we will set the SIP_NEEDREINVITE flag so that we can send a REINVITE as soon
as it is allowed.
The second part of the changes here is for Asterisk to accept UPDATE requests
that have connected line changes. Since we are not fully supporting RFC 3311,
Asterisk will NOT place the UPDATE method in Allow headers it sends. Instead,
if you are communicating with what you know to be another Asterisk box, you may
set the rpid_update parameter in sip.conf so that we will send UPDATEs to that
Asterisk box. When we send a connected line update, we set a custom header
called "X-Asterisk-rpid-update."
On the receiving end, if Asterisk receives an UPDATE that does not have the
"X-Asterisk-rpid-update" header present, then Asterisk will respond with a 501
since media-changing UPDATEs are not supported. We should never get such
UPDATEs, since as was stated earlier, Asterisk does not put UPDATE in its Allow
header. If the custom header is present in the received UPDATE, though, then we
will check the incoming request for connected line updates and queue the update
on the channel where the change occurred.
ABE-1840
ABE-1822
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Up to now, cdr_custom would only accept a single filename/format from
cdr_custom.conf. This change allows you to specify multiple filename
& format directives.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009) | 2 lines
Fix some spelling fail.
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Select what to do with outgoing COLP information on this port.
0 - Send out COLP information unaltered. (default)
1 - Force COLP to restricted on all outgoing COLP information.
2 - Do not send COLP information.
outgoing_colp=0
Also fixed sending the EctInform message so it always has the
required redirectionNumber parameter when the status is active.
JIRA ABE-1853
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May 2009) | 7 lines
Make absolute paths for logger channels work properly
(Note: This is not a new feature, it was previously undocumented and broken.)
The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf.
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This configuration file was changed to ensure that only one console channel driver
(chan_oss) is loaded by default, but the change would only work if chan_console
was not built. Now it will work as expected; if chan_alsa or chan_console are built
and installed, they will not be loaded unless explicity requested.
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In discussions today at the Europe Asterisk Developer Meet-Up, we determined that
the event_log was used in only 9 places in the entire tree, and really was not needed
at all. The users have been converted to use LOG_NOTICE, or the messages have been
removed since other messages were already in place that provided the same information.
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chan_sip allows for outbound TLS connections, but does not allow the user to specify what protocol to use (default was SSLv2, and still is if this new option is not specified). This patch lets the user pick the SSL/TLS client method for outbound connections in sip.
(closes issue #14770)
Reported by: TheOldSaint
(closes issue #14768)
Reported by: TheOldSaint
Review: http://reviewboard.digium.com/r/240/
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ast_tls_read_conf() is a new api call for handling SSL/TLS options across all conf files. Before this change, SSL/TLS options were not consistent. http.conf and manager.conf required the 'ssl' prefix while sip.conf used options with the 'tls' prefix. While the options had different names in different conf files, they all did the exact same thing. Now, instead of mixing 'ssl' or 'tls' prefixes to do the same thing depending on what conf file you're in, all SSL/TLS options use the 'tls' prefix. For example. 'sslenable' in http.conf and manager.conf is now 'tlsenable' which matches what already existed in sip.conf. Since this has the potential to break backwards compatibility, previous options containing the 'ssl' prefix still work, but they are no longer documented in the sample.conf files. The change is noted in the CHANGES file though.
Review: http://reviewboard.digium.com/r/237/
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The option to choose which connected line header to
use is not 'rpid_header' but 'sendrpid'
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Adds option to specify a private key .pem file when configuring TLS or SSL in AMI, HTTP, and SIP. Before this, the certificate file was used for both the public and private key. It is possible for this file to hold both, but most configurations allow for a separate private key file to be specified. Clarified in .conf files how these options are to be used. The current conf files do not explain how the private key is handled at all, so without knowledge of Asterisk's TLS implementation, it would be hard to know for sure what was going on or how to set it up.
Review: http://reviewboard.digium.com/r/234/
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This change adds the following features to chan_misdn:
* CCBS/CCNR Party A support for PTMP and PTP modes.
* Enhances COLP support for call diversion and explicit call transfer.
These enhanced features require a modified version of mISDN.
The latest modified mISDN v1.1.x based version is available at:
http://svn.digium.com/svn/thirdparty/mISDN/trunk
http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
Taged versions of the modified mISDN code are available under:
http://svn.digium.com/svn/thirdparty/mISDN/tags
http://svn.digium.com/svn/thirdparty/mISDNuser/tags
Review: http://reviewboard.digium.com/r/218/
Merged from team/rmudgett/misdn_facility branch.
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This is the companion commit to libpri r732. Service messages are now supported
for switch types 4ess/5ess. A new option service_message_support has been added
to chan_dahdi.conf and is noted in the sample config file. The service message
support is turned off by default. The current implementation relies on AstDB
to keep track of channel state, which allows the statuses to be preserved
across Asterisk restarts. Below is a description of the storage format.
The state and reason for the service state are in the form <state>:<reason>,
where:
<state> ::= { 'O' } // 'O' – Out Of Service
<reason> ::= { '0' | '1' | '2' | '3' }, where:
'0' – No reason (backwards compatibility)
'1' – NEAR END
'2' – FAR END
'3' – both NEAR and FAR END
The new CLI commands to handle channel service state are:
pri service disable channel <chan>
pri service enable channel <chan>
Many people contributed to the development of this functionality. Because I
entered at the very end I do not know the exact history. Special thanks to
all who moved the bug forward one way or another:
cmaj, PCadach, markster, mattf, drmac, MikeJ, serge-v, murf, kanelbullar, Seb7,
tilghman, lmadsen, and especially dhubbard (he answered lots of my questions
and did a large portion of the work)
(closes issue #3450)
Reported by: cmaj
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much better solution will be used
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This allows for you to change the From header for outgoing MWI
NOTIFY requests. Prior to this, the best you could do was to
set a callerid in the general section of sip.conf. The problem
was that this was used for all outbound requests, not just
MWI NOTIFY requests.
AST-201
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events, so that they can be fed into external processes for analysis and possible mitigation efforts
(inspired by this evening's Toronto Asterisk Users Group meeting and previous dicussions amongst various community members)
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