Added a new module, res_phoneprov, which allows auto-provisioning of phones
based on configuration templates that use Asterisk dialplan function and variable substitution. It should be possible to create phone profiles and templates that work for the majority of phones provisioned over http. It is currently only intended to provision a single user account per phone. An example profile and set of templates for Polycom phones is provided. NOTE: Polycom firmware is not included, but should be placed in AST_DATA_DIR/phoneprov/configs to match up with the included templates. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97634 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
parent
46f4c8946f
commit
3570ad103d
8
CHANGES
8
CHANGES
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@ -436,3 +436,11 @@ Miscellaneous
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* Added a new codec translation module, codec_resample, which re-samples
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signed linear audio between 8 kHz and 16 kHz to help support wideband
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codecs.
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* Added a new module, res_phoneprov, which allows auto-provisioning of phones
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based on configuration templates that use Asterisk dialplan function and
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variable substitution. It should be possible to create phone profiles and
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templates that work for the majority of phones provisioned over http. It
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is currently only intended to provision a single user account per phone.
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An example profile and set of templates for Polycom phones is provided.
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NOTE: Polycom firmware is not included, but should be placed in
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AST_DATA_DIR/phoneprov/configs to match up with the included templates.
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4
Makefile
4
Makefile
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@ -450,6 +450,10 @@ datafiles: _all
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# Should static HTTP be installed during make samples or even with its own target ala
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# webvoicemail? There are portions here that *could* be customized but might also be
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# improved a lot. I'll put it here for now.
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mkdir -p $(DESTDIR)$(ASTDATADIR)/phoneprov
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for x in phoneprov/*; do \
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$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTDATADIR)/phoneprov ; \
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done
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mkdir -p $(DESTDIR)$(ASTDATADIR)/static-http
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for x in static-http/*; do \
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$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTDATADIR)/static-http ; \
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@ -18,6 +18,9 @@ autoload=yes
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;preload => res_odbc.so
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;preload => res_config_odbc.so
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;
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; res_phoneprov requires func_strings.so to be loaded:
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preload => func_strings.so
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;
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; Uncomment the following if you wish to use the Speech Recognition API
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;preload => res_speech.so
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;
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@ -0,0 +1,55 @@
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[general]
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;serveraddr=192.168.1.1 ; Address to send to the phone to use as server address.
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serveriface=eth0 ; Same as above, except an ethernet interface.
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; Useful for when the interface uses DHCP.
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; There is no default for either of the above, and only one should be set.
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serverport=5060 ; Port to send to the phone to use as server port. Default is 5060.
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default_profile=polycom ; The default profile to use if none specified in users.conf
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; You can define profiles for different phones specifying what files to register
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; with the provisioning server. You can define either static files, or dynamically
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; generated files that can have dynamic names and point to templates that variables
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; can be substituted into. You can also set arbitrary variables for the profiles
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; templates to have access to. Example:
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;[example]
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;mime_type => application/octet-stream
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;static_file => example/firmware
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;static_file => example/default.cfg,text/xml
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;${TOUPPER(${MAC})}.cfg => templates/example-mac.cfg
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;setvar => DB_CIDNAME=${ODBC_CID_NAME_LOOKUP(${USERNAME})}
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; Dynamically generated files have a filename registered with variable substitution
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; with variables obtained while reading users.conf.
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; Built in variables and the options in users.conf that they come from
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; MAC (macaddress)
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; USERNAME (username)
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; DISPLAY_NAME (fullname)
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; SECRET (secret)
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; LABEL (label)
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; CALLERID (cid_number)
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; VOCIEMAIL_EXTEN (vmexten)
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; EXTENSION_LENGTH (localextenlength)
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; Built-in variables and the options in phoneprov.conf that they come from
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; SERVER (server)
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; SERVER_PORT (serverport)
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[polycom]
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staticdir => configs/ ; Sub directory of AST_DATA_DIR/phoneprov that static files reside
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; in. This allows a request to /phoneprov/sip.cfg to pull the file
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; from /phoneprov/configs/sip.cfg
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mime_type => text/xml ; Default mime type to use if one isn't specified or the
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; extension isn't recognized
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static_file => bootrom.ld,application/octet-stream ; Static files the phone will download
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static_file => bootrom.ver,plain/text ; static_file => filename,mime-type
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static_file => sip.ld,application/octet-stream
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static_file => sip.ver,plain/text
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static_file => sip.cfg
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static_file => custom.cfg
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${TOLOWER(${MAC})}.cfg => 000000000000.cfg ; Dynamically generated files.
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${TOLOWER(${MAC})}-phone.cfg => 000000000000-phone.cfg ; (relative to AST_DATA_DIR/phoneprov)
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config/${TOLOWER(${MAC})} => polycom.xml ; Dynamic Filename => template file
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${TOLOWER(${MAC})}-directory.xml => 000000000000-directory.xml
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setvar => CUSTOM_CONFIG=/var/lib/asterisk/phoneprov/configs/custom.cfg ; Custom variable
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@ -32,7 +32,7 @@
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\author{Asterisk Development Team \\ Asterisk.org}
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\title{Asterisk Reference Information \\ Version ASTERISKVERSION}
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\title{Asterisk Reference Information \\ Version }
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\begin{document}
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\maketitle
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@ -132,6 +132,9 @@ reference purposes.
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\section{Queue Logs}
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\input{queuelog.tex}
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\chapter{Phone Provisioning}
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\input{phoneprov.tex}
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\chapter{Development}
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\section{Backtrace}
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\input{backtrace.tex}
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@ -0,0 +1,301 @@
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\section{Introduction}
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Asterisk includes basic phone provisioning support through the res\_phoneprov module. The
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current implementation is based on a templating system using Asterisk dialplan function
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and variable substitution and obtains information to substitute into those templates from
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\path{phoneprov.conf} and \path{users.conf}. A profile and set of templates is provided
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for provisioning Polycom phones. Note that res\_phoneprov is currently limited to
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provisioning a single user per device.
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\section{Configuration of phoneprov.conf}
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The configuration file, \path{phoneprov.conf}, is used to set up the built-in variables
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SEVER and SERVER\_PORT, to define a default phone profile to use, and to define different
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phone profiles available for provisioning.
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\subsection{The [general] section}
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Below is a sample of the general section of \path{phoneprov.conf}:
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\begin{astlisting}
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\begin{verbatim}
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[general]
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;serveriface=eth0
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serveraddr=192.168.1.1
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serverport=5060
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default_profile=polycom
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\end{verbatim}
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\end{astlisting}
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There are two choices for setting the SERVER variable. If the IP address of the server is
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known, or the hostname resolvable by the phones, the appropriate \textbf{serveraddr}
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value should be set. Alternatively, the network interface that the server listens on can
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be set by specifying a \textbf{serveriface} and SERVER will be set to the IP address of
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that interface. Only one of these options should be set.
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The SERVER\_PORT variable is set by setting the \textbf{serverport}. If serverport is
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not specified, it is set to a default value of 5060.
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Any user set for auto-provisioning in users.conf without a specified profile will be
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assumed to belong to the profile set with \textbf{default\_profile}.
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\subsection{Creating phone profiles}
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A phone profile is basically a list of files that a particular group of phones needs to
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function. For most phone types there are files that are identical for all phones
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(firmware, for instance) as well as a configuration file that is specific to individual
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phones. res\_phoneprov breaks these two groups of files into static files and dynamic
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files, respectively. A sample profile:
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\begin{astlisting}
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\begin{verbatim}
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[polycom]
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staticdir => configs/
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mime_type => text/xml
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setvar => CUSTOM_CONFIG=/var/lib/asterisk/phoneprov/configs/custom.cfg
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static_file => bootrom.ld,application/octet-stream
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static_file => bootrom.ver,plain/text
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static_file => sip.ld,application/octet-stream
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static_file => sip.ver,plain/text
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static_file => sip.cfg
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static_file => custom.cfg
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${TOLOWER(${MAC})}.cfg => 000000000000.cfg
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${TOLOWER(${MAC})}-phone.cfg => 000000000000-phone.cfg
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config/${TOLOWER(${MAC})} => polycom.xml
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${TOLOWER(${MAC})}-directory.xml => 000000000000-directory.xml
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\end{verbatim}
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\end{astlisting}
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A \textbf{static\_file} is set by specifying the file name, relative to
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\path{AST\_DATA\_DIR/phoneprov}. The mime-type of the file can optionally be specified
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after a comma. If \textbf{staticdir} is set, all static files will be relative to the
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subdirectory of AST\_DATA\_DIR/phoneprov specified.
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Since phone-specific config files generally have file names based on phone-specifc data,
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dynamic filenames in res\_phoneprov can be defined with Asterisk dialplan function and
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variable substitution. In the above example, \$\{TOLOWER(\$\{MAC\})\}.cfg $\Rightarrow$
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000000000000.cfg would define a relative URI to be served that matches the format of
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MACADDRESS.cfg, all lower case. A request for that file would then point to the template
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found at AST\_DATA\_DIR/phoneprov/000000000000.cfg. The template can be followed by a
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comma and mime-type. Notice that the dynamic filename (URI) can contain contain
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directories. Since these files are dynamically generated, the config file itself does not
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reside on the filesystem--only the template. To view the generated config file, open it
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in a web browser. If the config file is XML, Firefox should display it. Some browsers
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will require viewing the source of the page requested.
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A default mime-type for the profile can be defined by setting \textbf{mime-type}. If a
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custom variable is required for a template, it can be specified with \textbf{setvar}.
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Variable substitution on this value is done while building the route list, so
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\$\{USERNAME\} would expand to the username of the users.conf user that registers the
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dynamic filename.
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NOTE: Any dialplan function that is used for generation of dynamic file names MUST be
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loaded before res\_phoneprov. Add "preload $\Rightarrow$ modulename.so" to
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\path{modules.conf} for required functions. In the example above, "preload $\Rightarrow$
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func\_strings.so" would be required.
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\section{Configuration of users.conf}
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The asterisk-gui sets up extensions, SIP/IAX2 peers, and a host of other settings.
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User-specific settings are stored in users.conf. If the asterisk-gui is not being used,
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manual entries to users.conf can be made.
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\subsection{The [general] section}
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||||
There are only two settings in the general section of \path{users.conf} that apply to
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phone provisioning: localextenlength which maps to template variable EXTENSION\_LENGTH
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and \textbf{vmexten} which maps to the VOICEMAIL\_EXTEN variable.
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\subsection{Invdividual Users}
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To enable auto-provisioning of a phone, the user in \path{users.conf} needs to have:
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\begin{astlisting}
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\begin{verbatim}
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||||
...
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||||
autoprov=yes
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macaddress=deadbeef4dad
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profile=polycom
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\end{verbatim}
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\end{astlisting}
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||||
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||||
The profile is optional if a \textbf{default\_profile} is set in \path{phoneprov.conf}.
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The following is a sample users.conf entry, with the template variables commented next to
|
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the settings:
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||||
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||||
\begin{astlisting}
|
||||
\begin{verbatim}
|
||||
[6001]
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||||
callwaiting = yes
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||||
context = numberplan-custom-1
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hasagent = no
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||||
hasdirectory = yes
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hasiax = no
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hasmanager = no
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hassip = yes
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hasvoicemail = yes
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host = dynamic
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mailbox = 6001
|
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threewaycalling = yes
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deletevoicemail = no
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autoprov = yes
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profile = polycom
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canreinvite = no
|
||||
nat = no
|
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fullname = User Two ; ${DISPLAY_NAME}
|
||||
secret = test ; ${SECRET}
|
||||
username = 6001 ; ${USERNAME}
|
||||
macaddress = deadbeef4dad ; ${MAC}
|
||||
label = 6001 ; ${LABEL}
|
||||
cid_number = 6001 ; ${CALLERID}
|
||||
\end{verbatim}
|
||||
\end{astlisting}
|
||||
|
||||
The variables above, are the user-specfic variables that can be substituted into dynamic
|
||||
filenames and config templates.
|
||||
|
||||
\section{Templates}
|
||||
|
||||
Configuration templates are a generic way to configure phones with text-based
|
||||
configuration files. Templates can use any loaded dialplan function and all of the
|
||||
variables created by \path{phoneprov.conf} and \path{users.conf}. A short example is the
|
||||
included 000000000000.cfg Polycom template:
|
||||
|
||||
\begin{astlisting}
|
||||
\begin{verbatim}
|
||||
<?xml version="1.0" standalone="yes"?>
|
||||
<APPLICATION
|
||||
APP_FILE_PATH="sip.ld"
|
||||
CONFIG_FILES="${IF($[${STAT(e|${CUSTOM_CONFIG})}] ? "custom.cfg,
|
||||
")}config/${TOLOWER(${MAC})}, sip.cfg"
|
||||
MISC_FILES="" LOG_FILE_DIRECTORY=""
|
||||
/>
|
||||
\end{verbatim}
|
||||
\end{astlisting}
|
||||
|
||||
This template uses dialplan functions, expressions, and a couple of variables to generate
|
||||
a config file to instruct the Polycom where to pull other needed config files. If a phone
|
||||
with MAC address 0xDEADBEEF4DAD requests this config file, and the filename that is
|
||||
stored in variable CUSTOM\_CONFIG does not exist, then the generated output would be:
|
||||
|
||||
\begin{astlisting}
|
||||
\begin{verbatim}
|
||||
<?xml version="1.0" standalone="yes"?>
|
||||
<APPLICATION
|
||||
APP_FILE_PATH="sip.ld"
|
||||
CONFIG_FILES="config/deadbeef4dad, sip.cfg"
|
||||
MISC_FILES="" LOG_FILE_DIRECTORY=""
|
||||
/>
|
||||
\end{verbatim}
|
||||
\end{astlisting}
|
||||
|
||||
The Polycom phone would then download both sip.cfg (which would be registered in
|
||||
\path{phoneprov.conf} as a static file) and config/deadbeef4dad (which would be
|
||||
registered as a dynamic file pointing to another template, polycom.xml).
|
||||
|
||||
res\_phoneprov also registers its own dialplan function: PP\_EACH\_USER. This function
|
||||
was designed to be able to print out a particular string for each user that
|
||||
res\_phoneprov knows about. An example use of this function is the template for a Polycom
|
||||
contact directory:
|
||||
|
||||
\begin{astlisting}
|
||||
\begin{verbatim}
|
||||
<?xml version="1.0" standalone="yes"?>
|
||||
<directory>
|
||||
<item_list>
|
||||
${PP_EACH_USER(<item><fn>%{DISPLAY_NAME}</fn><ct>%{CALLERID}</ct><bw>1</bw></item>|${MAC})}
|
||||
</item_list>
|
||||
</directory>
|
||||
\end{verbatim}
|
||||
\end{astlisting}
|
||||
|
||||
PP\_EACH\_USER takes two arguments. The first is the string to be printed for each user.
|
||||
Any variables that are to be substituted need to be in the format \%\{VARNAME\} so that
|
||||
Asterisk doesn't try to substitute the variable immediately before it is passed to
|
||||
PP\_EACH\_USER. The second, optional, argument is a MAC address to exclude from the list
|
||||
iterated over (so, in this case, a phone won't be listed in its own contact directory).
|
||||
|
||||
\section{Putting it all together}
|
||||
|
||||
Make sure that \path{manager.conf} has:
|
||||
|
||||
\begin{astlisting}
|
||||
\begin{verbatim}
|
||||
[general]
|
||||
enabled = yes
|
||||
webenabled = yes
|
||||
\end{verbatim}
|
||||
\end{astlisting}
|
||||
|
||||
and that \path{http.conf} has:
|
||||
|
||||
\begin{astlisting}
|
||||
\begin{verbatim}
|
||||
[general]
|
||||
enabled = yes
|
||||
bindaddr = 192.168.1.1 ; Your IP here ;-)
|
||||
bindport = 8088 ; Or port 80 if it is the only http server running on the machine
|
||||
\end{verbatim}
|
||||
\end{astlisting}
|
||||
|
||||
With \path{phoneprov.conf} and \path{users.conf} in place, start Astersik. From the CLI,
|
||||
type "http show status". An example output:
|
||||
\begin{astlisting}
|
||||
\begin{verbatim}
|
||||
HTTP Server Status:
|
||||
Prefix: /asterisk
|
||||
Server Enabled and Bound to 192.168.1.1:8088
|
||||
|
||||
Enabled URI's:
|
||||
/asterisk/httpstatus => Asterisk HTTP General Status
|
||||
/asterisk/phoneprov/... => Asterisk HTTP Phone Provisioning Tool
|
||||
/asterisk/manager => HTML Manager Event Interface
|
||||
/asterisk/rawman => Raw HTTP Manager Event Interface
|
||||
/asterisk/static/... => Asterisk HTTP Static Delivery
|
||||
/asterisk/mxml => XML Manager Event Interface
|
||||
|
||||
Enabled Redirects:
|
||||
None.
|
||||
|
||||
POST mappings:
|
||||
None.
|
||||
\end{verbatim}
|
||||
\end{astlisting}
|
||||
|
||||
There should be a phoneprov URI listed. Next, from the CLI, type "phoneprov show routes"
|
||||
and verify that the information there is correct. An example output for Polycom phones
|
||||
woud look like:
|
||||
|
||||
\begin{astlisting}
|
||||
\begin{verbatim}
|
||||
Static routes
|
||||
|
||||
Relative URI Physical location
|
||||
sip.ver configs/sip.ver
|
||||
sip.ld configs/sip.ld
|
||||
bootrom.ver configs/bootrom.ver
|
||||
sip.cfg configs/sip.cfg
|
||||
bootrom.ld configs/bootrom.ld
|
||||
custom.cfg configs/custom.cfg
|
||||
|
||||
Dynamic routes
|
||||
|
||||
Relative URI Template
|
||||
deadbeef4dad.cfg 000000000000.cfg
|
||||
deadbeef4dad-directory.xml 000000000000-directory.xml
|
||||
deadbeef4dad-phone.cfg 000000000000-phone.cfg
|
||||
config/deadbeef4dad polycom.xml
|
||||
\end{verbatim}
|
||||
\end{astlisting}
|
||||
|
||||
With the above examples, the phones would be pointed to
|
||||
\url{http://192.168.1.1:8080/asterisk/phoneprov} for pulling config files. Templates
|
||||
would all be placed in AST\_DATA\_DIR/phoneprov and static files would be placed in
|
||||
AST\_DATA\_DIR/phoneprov/configs. Examples of valid URIs would be:
|
||||
|
||||
\begin{itemize}
|
||||
\item http://192.168.1.1:8080/asterisk/phoneprov/sip.cfg
|
||||
\item http://192.168.1.1:8080/asterisk/phoneprov/deadbeef4dad.cfg
|
||||
\item http://192.168.1.1:8080/asterisk/phoneprov/config/deadbeef4dad
|
||||
\end{itemize}
|
||||
|
|
@ -30,6 +30,7 @@
|
|||
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
||||
|
||||
#include <regex.h>
|
||||
#include <ctype.h>
|
||||
|
||||
#include "asterisk/module.h"
|
||||
#include "asterisk/channel.h"
|
||||
|
@ -800,6 +801,40 @@ static struct ast_custom_function keypadhash_function = {
|
|||
.desc = "Example: ${KEYPADHASH(Les)} returns \"537\"\n",
|
||||
};
|
||||
|
||||
static int string_toupper(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
|
||||
{
|
||||
char *bufptr = buf, *dataptr = data;
|
||||
|
||||
while ((bufptr < buf + len - 1) && (*bufptr++ = toupper(*dataptr++)));
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static struct ast_custom_function toupper_function = {
|
||||
.name = "TOUPPER",
|
||||
.synopsis = "Convert the string to upper case.",
|
||||
.syntax = "TOUPPER(<string>)",
|
||||
.read = string_toupper,
|
||||
.desc = "Example: ${TOUPPER(Example)} returns \"EXAMPLE\"\n",
|
||||
};
|
||||
|
||||
static int string_tolower(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
|
||||
{
|
||||
char *bufptr = buf, *dataptr = data;
|
||||
|
||||
while ((bufptr < buf + len - 1) && (*bufptr++ = tolower(*dataptr++)));
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static struct ast_custom_function tolower_function = {
|
||||
.name = "TOLOWER",
|
||||
.synopsis = "Convert the string to lower case.",
|
||||
.syntax = "TOLOWER(<string>)",
|
||||
.read = string_tolower,
|
||||
.desc = "Example: ${TOLOWER(Example)} returns \"example\"\n",
|
||||
};
|
||||
|
||||
static int unload_module(void)
|
||||
{
|
||||
int res = 0;
|
||||
|
@ -818,6 +853,8 @@ static int unload_module(void)
|
|||
res |= ast_custom_function_unregister(&hashkeys_function);
|
||||
res |= ast_custom_function_unregister(&hash_function);
|
||||
res |= ast_unregister_application(app_clearhash);
|
||||
res |= ast_custom_function_unregister(&toupper_function);
|
||||
res |= ast_custom_function_unregister(&tolower_function);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
@ -840,6 +877,8 @@ static int load_module(void)
|
|||
res |= ast_custom_function_register(&hashkeys_function);
|
||||
res |= ast_custom_function_register(&hash_function);
|
||||
res |= ast_register_application(app_clearhash, exec_clearhash, syn_clearhash, desc_clearhash);
|
||||
res |= ast_custom_function_register(&toupper_function);
|
||||
res |= ast_custom_function_register(&tolower_function);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
|
|
@ -41,6 +41,7 @@ struct ast_tm {
|
|||
};
|
||||
|
||||
struct ast_tm *ast_localtime(const struct timeval *timep, struct ast_tm *p_tm, const char *zone);
|
||||
void ast_get_dst_info(const time_t * const timep, int *dst_enabled, time_t *dst_start, time_t *dst_end, int *gmt_off, const char * const zone);
|
||||
struct timeval ast_mktime(struct ast_tm * const tmp, const char *zone);
|
||||
int ast_strftime(char *buf, size_t len, const char *format, const struct ast_tm *tm);
|
||||
|
||||
|
|
|
@ -39,11 +39,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
|
|||
#include "asterisk/lock.h"
|
||||
#include "asterisk/srv.h"
|
||||
|
||||
struct my_ifreq {
|
||||
char ifrn_name[IFNAMSIZ]; /* Interface name, e.g. "eth0", "ppp0", etc. */
|
||||
struct sockaddr_in ifru_addr;
|
||||
};
|
||||
|
||||
/* Free HA structure */
|
||||
void ast_free_ha(struct ast_ha *ha)
|
||||
{
|
||||
|
|
|
@ -1143,6 +1143,121 @@ struct ast_tm *ast_localtime(const struct timeval *timep, struct ast_tm *tmp, co
|
|||
return sp ? localsub(timep, 0L, tmp, sp) : NULL;
|
||||
}
|
||||
|
||||
/*
|
||||
** This function provides informaton about daylight savings time
|
||||
** for the given timezone. This includes whether it can determine
|
||||
** if daylight savings is used for this timezone, the UTC times for
|
||||
** when daylight savings transitions, and the offset in seconds from
|
||||
** UTC.
|
||||
*/
|
||||
|
||||
void ast_get_dst_info(const time_t * const timep, int *dst_enabled, time_t *dst_start, time_t *dst_end, int *gmt_off, const char * const zone)
|
||||
{
|
||||
int i;
|
||||
int transition1 = -1;
|
||||
int transition2 = -1;
|
||||
time_t seconds;
|
||||
int bounds_exceeded = 0;
|
||||
time_t t = *timep;
|
||||
const struct state *sp;
|
||||
|
||||
if (NULL == dst_enabled)
|
||||
return;
|
||||
*dst_enabled = 0;
|
||||
|
||||
if (NULL == dst_start || NULL == dst_end || NULL == gmt_off)
|
||||
return;
|
||||
|
||||
*gmt_off = 0;
|
||||
|
||||
sp = ast_tzset(zone);
|
||||
if (NULL == sp)
|
||||
return;
|
||||
|
||||
/* If the desired time exceeds the bounds of the defined time transitions
|
||||
* then give give up on determining DST info and simply look for gmt offset
|
||||
* This requires that I adjust the given time using increments of Gregorian
|
||||
* repeats to place the time within the defined time transitions in the
|
||||
* timezone structure.
|
||||
*/
|
||||
if ((sp->goback && t < sp->ats[0]) ||
|
||||
(sp->goahead && t > sp->ats[sp->timecnt - 1])) {
|
||||
time_t tcycles;
|
||||
int_fast64_t icycles;
|
||||
|
||||
if (t < sp->ats[0])
|
||||
seconds = sp->ats[0] - t;
|
||||
else seconds = t - sp->ats[sp->timecnt - 1];
|
||||
--seconds;
|
||||
tcycles = seconds / YEARSPERREPEAT / AVGSECSPERYEAR;
|
||||
++tcycles;
|
||||
icycles = tcycles;
|
||||
if (tcycles - icycles >= 1 || icycles - tcycles >= 1)
|
||||
return;
|
||||
seconds = icycles;
|
||||
seconds *= YEARSPERREPEAT;
|
||||
seconds *= AVGSECSPERYEAR;
|
||||
if (t < sp->ats[0])
|
||||
t += seconds;
|
||||
else
|
||||
t -= seconds;
|
||||
|
||||
if (t < sp->ats[0] || t > sp->ats[sp->timecnt - 1])
|
||||
return; /* "cannot happen" */
|
||||
|
||||
bounds_exceeded = 1;
|
||||
}
|
||||
|
||||
if (sp->timecnt == 0 || t < sp->ats[0]) {
|
||||
/* I have no transition times or I'm before time */
|
||||
*dst_enabled = 0;
|
||||
/* Find where I can get gmtoff */
|
||||
i = 0;
|
||||
while (sp->ttis[i].tt_isdst)
|
||||
if (++i >= sp->typecnt) {
|
||||
i = 0;
|
||||
break;
|
||||
}
|
||||
*gmt_off = sp->ttis[i].tt_gmtoff;
|
||||
return;
|
||||
}
|
||||
|
||||
for (i = 1; i < sp->timecnt; ++i) {
|
||||
if (t < sp->ats[i]) {
|
||||
transition1 = sp->types[i - 1];
|
||||
transition2 = sp->types[i];
|
||||
break;
|
||||
}
|
||||
}
|
||||
/* if I found transition times that do not bounded the given time and these correspond to
|
||||
or the bounding zones do not reflect a changes in day light savings, then I do not have dst active */
|
||||
if (i >= sp->timecnt || 0 > transition1 || 0 > transition2 ||
|
||||
(sp->ttis[transition1].tt_isdst == sp->ttis[transition2].tt_isdst)) {
|
||||
*dst_enabled = 0;
|
||||
*gmt_off = sp->ttis[sp->types[sp->timecnt -1]].tt_gmtoff;
|
||||
} else {
|
||||
/* I have valid daylight savings information. */
|
||||
if(sp->ttis[transition2].tt_isdst)
|
||||
*gmt_off = sp->ttis[transition1].tt_gmtoff;
|
||||
else
|
||||
*gmt_off = sp->ttis[transition2].tt_gmtoff;
|
||||
|
||||
/* If I adjusted the time earlier, indicate that the dst is invalid */
|
||||
if (!bounds_exceeded) {
|
||||
*dst_enabled = 1;
|
||||
/* Determine which of the bounds is the start of daylight savings and which is the end */
|
||||
if(sp->ttis[transition2].tt_isdst) {
|
||||
*dst_start = sp->ats[i];
|
||||
*dst_end = sp->ats[i -1];
|
||||
} else {
|
||||
*dst_start = sp->ats[i -1];
|
||||
*dst_end = sp->ats[i];
|
||||
}
|
||||
}
|
||||
}
|
||||
return;
|
||||
}
|
||||
|
||||
/*
|
||||
** gmtsub is to gmtime as localsub is to localtime.
|
||||
*/
|
||||
|
|
|
@ -0,0 +1,6 @@
|
|||
<?xml version="1.0" standalone="yes"?>
|
||||
<directory>
|
||||
<item_list>
|
||||
${PP_EACH_USER(<item><fn>%{DISPLAY_NAME}</fn><ct>%{CALLERID}</ct><bw>1</bw></item>|${MAC})}
|
||||
</item_list>
|
||||
</directory>
|
|
@ -0,0 +1 @@
|
|||
<?xml version="1.0" standalone="yes"?>
|
|
@ -0,0 +1,2 @@
|
|||
<?xml version="1.0" standalone="yes"?>
|
||||
<APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="${IF($[${STAT(e|${CUSTOM_CONFIG})}] ? "custom.cfg, ")}config/${TOLOWER(${MAC})}, sip.cfg" MISC_FILES="" LOG_FILE_DIRECTORY=""/>
|
|
@ -0,0 +1,37 @@
|
|||
<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
|
||||
<phone1>
|
||||
<reg reg.1.displayName="${DISPLAY_NAME}" reg.1.address="${USERNAME}" reg.1.label="${LABEL}" reg.1.type="private" reg.1.thirdPartyName="" reg.1.auth.userId="${USERNAME}" reg.1.auth.password="${SECRET}" reg.1.server.1.address="${SERVER}" reg.1.server.1.port="${SERVER_PORT}" reg.1.server.1.transport="DNSnaptr" reg.1.server.1.expires="300" reg.1.server.1.register="1" reg.1.server.1.retryTimeOut="" reg.1.server.1.retryMaxCount="" reg.1.server.1.expires.lineSeize="" reg.1.acd-login-logout="0" reg.1.acd-agent-available="0" reg.1.ringType="2" reg.1.lineKeys="2" reg.1.callsPerLineKey="1"/>
|
||||
<call>
|
||||
<donotdisturb call.donotdisturb.perReg="1" />
|
||||
<autoOffHook call.autoOffHook.1.enabled="0" call.autoOffHook.1.contact="" call.autoOffHook.2.enabled="0" call.autoOffHook.2.contact="" call.autoOffHook.3.enabled="0" call.autoOffHook.3.contact="" call.autoOffHook.4.enabled="0" call.autoOffHook.4.contact="" call.autoOffHook.5.enabled="0" call.autoOffHook.5.contact="" call.autoOffHook.6.enabled="0" call.autoOffHook.6.contact=""/>
|
||||
<serverMissedCall call.serverMissedCall.1.enabled="0" call.serverMissedCall.2.enabled="0" call.serverMissedCall.3.enabled="0" call.serverMissedCall.4.enabled="0" call.serverMissedCall.5.enabled="0" call.serverMissedCall.6.enabled="0"/>
|
||||
</call>
|
||||
<divert divert.1.contact="" divert.1.autoOnSpecificCaller="1" divert.1.sharedDisabled="1" divert.2.contact="" divert.2.autoOnSpecificCaller="1" divert.2.sharedDisabled="1" divert.3.contact="" divert.3.autoOnSpecificCaller="1" divert.3.sharedDisabled="1" divert.4.contact="" divert.4.autoOnSpecificCaller="1" divert.4.sharedDisabled="1" divert.5.contact="" divert.5.autoOnSpecificCaller="1" divert.5.sharedDisabled="1" divert.6.contact="" divert.6.autoOnSpecificCaller="1" divert.6.sharedDisabled="1">
|
||||
<fwd divert.fwd.1.enabled="1" divert.fwd.2.enabled="1" divert.fwd.3.enabled="1" divert.fwd.4.enabled="1" divert.fwd.5.enabled="1" divert.fwd.6.enabled="1"/>
|
||||
<busy divert.busy.1.enabled="1" divert.busy.1.contact="" divert.busy.2.enabled="1" divert.busy.2.contact="" divert.busy.3.enabled="1" divert.busy.3.contact="" divert.busy.4.enabled="1" divert.busy.4.contact="" divert.busy.5.enabled="1" divert.busy.5.contact="" divert.busy.6.enabled="1" divert.busy.6.contact=""/>
|
||||
<noanswer divert.noanswer.1.enabled="1" divert.noanswer.1.timeout="60" divert.noanswer.1.contact="" divert.noanswer.2.enabled="1" divert.noanswer.2.timeout="60" divert.noanswer.2.contact="" divert.noanswer.3.enabled="1" divert.noanswer.3.timeout="60" divert.noanswer.3.contact="" divert.noanswer.4.enabled="1" divert.noanswer.4.timeout="60" divert.noanswer.4.contact="" divert.noanswer.5.enabled="1" divert.noanswer.5.timeout="60" divert.noanswer.5.contact="" divert.noanswer.6.enabled="1" divert.noanswer.6.timeout="60" divert.noanswer.6.contact=""/>
|
||||
<dnd divert.dnd.1.enabled="0" divert.dnd.1.contact="" divert.dnd.2.enabled="0" divert.dnd.2.contact="" divert.dnd.3.enabled="0" divert.dnd.3.contact="" divert.dnd.4.enabled="0" divert.dnd.4.contact="" divert.dnd.5.enabled="0" divert.dnd.5.contact="" divert.dnd.6.enabled="0" divert.dnd.6.contact=""/>
|
||||
</divert>
|
||||
<dialplan
|
||||
dialplan.1.impossibleMatchHandling="2" dialplan.1.removeEndOfDial="0"
|
||||
/>
|
||||
<digitmap
|
||||
dialplan.1.digitmap="" dialplan.1.digitmap.timeOut="3"
|
||||
/>
|
||||
<routing>
|
||||
<server dialplan.1.routing.server.1.address="" dialplan.1.routing.server.1.port="" dialplan.2.routing.server.1.address="" dialplan.2.routing.server.1.port="" dialplan.3.routing.server.1.address="" dialplan.3.routing.server.1.port="" dialplan.4.routing.server.1.address="" dialplan.4.routing.server.1.port="" dialplan.5.routing.server.1.address="" dialplan.5.routing.server.1.port="" dialplan.6.routing.server.1.address="" dialplan.6.routing.server.1.port=""/>
|
||||
<emergency dialplan.1.routing.emergency.1.value="" dialplan.1.routing.emergency.1.server.1="" dialplan.2.routing.emergency.1.value="" dialplan.2.routing.emergency.1.server.1="" dialplan.3.routing.emergency.1.value="" dialplan.3.routing.emergency.1.server.1="" dialplan.4.routing.emergency.1.value="" dialplan.4.routing.emergency.1.server.1="" dialplan.5.routing.emergency.1.value="" dialplan.5.routing.emergency.1.server.1="" dialplan.6.routing.emergency.1.value="" dialplan.6.routing.emergency.1.server.1=""/>
|
||||
</routing>
|
||||
<msg msg.bypassInstantMessage="1">
|
||||
<mwi msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="${VOICEMAIL_EXTEN}" />
|
||||
</msg>
|
||||
<nat nat.ip="" nat.signalPort="" nat.mediaPortStart=""/>
|
||||
<user_preferences up.oneTouchVoiceMail="1" up.welcomeSoundEnabled="0" />
|
||||
<volume voice.volume.persist.handset="1" voice.volume.persist.headset="1" />
|
||||
<SNTP tcpIpApp.sntp.address="time" tcpIpApp.sntp.gmtOffset="-21600" />
|
||||
<HTTPD httpd.enabled="1" />
|
||||
<feature
|
||||
feature.1.name="presence" feature.1.enabled="1"
|
||||
feature.8.name="calllist-missed" feature.8.enabled="1"
|
||||
/>
|
||||
</phone1>
|
File diff suppressed because it is too large
Load Diff
Loading…
Reference in New Issue