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asterisk/configs/samples/phoneprov.conf.sample

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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
2008-01-09 21:37:26 +00:00
[general]
users.conf: Remove deprecated users.conf integration. users.conf was deprecated in Asterisk 21 and is now being removed for Asterisk 23, in accordance with the Asterisk deprecation policy. This consists of: * Removing integration with app_directory, app_voicemail, chan_dahdi, chan_iax2, and AMI. * users.conf was also partially used for res_phoneprov, and this remaining functionality is consolidated to a separate phoneprov_users.conf, used only by res_phoneprov. Resolves: #1292 UpgradeNote: users.conf has been removed and all channel drivers must be configured using their specific configuration files. The functionality previously in users.conf for res_phoneprov is now in phoneprov_users.conf.
2025-07-09 09:30:43 -04:00
; This section applies only to the default phoneprov_users.conf config provider
res_phoneprov: Refactor phoneprov to allow pluggable config providers This patch makes res_phoneprov more modular so other modules (like pjsip) can provide configuration information instead of res_phoneprov relying solely on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API is now exposed which allows config providers to register themselves, set defaults (server profile, etc) and add user extensions. * ast_phoneprov_provider_register registers the provider and provides callbacks for loading default settings and loading users. * ast_phoneprov_provider_unregister clears the defaults and users. * ast_phoneprov_add_extension should be called once for each user/extension by the provider's load_users callback to add them. * ast_phoneprov_delete_extension deletes one extension. * ast_phoneprov_delete_extensions deletes all extensions for the provider. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3970/ ........ Merged revisions 424963 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424964 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424965 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-09 17:46:23 +00:00
; embedded in res_phoneprov. Other providers may provide their own default settings.
Update res_phoneprov to default to setting the SERVER variable to the IP the HTTP request for the config came in on and the SERVER_PORT to the bindport setting in sip.conf. I've left in the ability to override these options, because I can't always guess how someone might decide to do something weird with what is available to them--although needing to is pretty unlikely. Documentation was updated to reflect preference for not setting serveraddr, serveriface, or serverport. Tested on Linux and OS X. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-17 03:09:32 +00:00
; The default behavior of res_phoneprov will be to set the SERVER template variable to
chan_sip: Remove deprecated module. ASTERISK-30297 Change-Id: Ic700168c80b68879d9cee8bb07afe2712fb17996
2022-11-28 13:05:21 -07:00
; the IP address that the phone uses to contact the provisioning server. Unless you have
; an unusual setup, you should not need to set serveraddr, serveriface, or serverport.
Update res_phoneprov to default to setting the SERVER variable to the IP the HTTP request for the config came in on and the SERVER_PORT to the bindport setting in sip.conf. I've left in the ability to override these options, because I can't always guess how someone might decide to do something weird with what is available to them--although needing to is pretty unlikely. Documentation was updated to reflect preference for not setting serveraddr, serveriface, or serverport. Tested on Linux and OS X. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-17 03:09:32 +00:00
;serveraddr=192.168.1.1 ; Override address to send to the phone to use as server address.
;serveriface=eth0 ; Same as above, except an ethernet interface.
Remove a bunch of trailing whitespace in preparation for reformatting/cleanup. Let's try that again, this time removing trailing whitespace and not leading whitespace. I can't believe no one noticed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 14:39:21 +00:00
; Useful for when the interface uses DHCP and the asterisk http
chan_sip: Remove deprecated module. ASTERISK-30297 Change-Id: Ic700168c80b68879d9cee8bb07afe2712fb17996
2022-11-28 13:05:21 -07:00
; server listens on a different IP than sip.
Update res_phoneprov to default to setting the SERVER variable to the IP the HTTP request for the config came in on and the SERVER_PORT to the bindport setting in sip.conf. I've left in the ability to override these options, because I can't always guess how someone might decide to do something weird with what is available to them--although needing to is pretty unlikely. Documentation was updated to reflect preference for not setting serveraddr, serveriface, or serverport. Tested on Linux and OS X. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-17 03:09:32 +00:00
;serverport=5060 ; Override port to send to the phone to use as server port.
users.conf: Remove deprecated users.conf integration. users.conf was deprecated in Asterisk 21 and is now being removed for Asterisk 23, in accordance with the Asterisk deprecation policy. This consists of: * Removing integration with app_directory, app_voicemail, chan_dahdi, chan_iax2, and AMI. * users.conf was also partially used for res_phoneprov, and this remaining functionality is consolidated to a separate phoneprov_users.conf, used only by res_phoneprov. Resolves: #1292 UpgradeNote: users.conf has been removed and all channel drivers must be configured using their specific configuration files. The functionality previously in users.conf for res_phoneprov is now in phoneprov_users.conf.
2025-07-09 09:30:43 -04:00
default_profile=polycom ; The default profile to use if none specified in phoneprov_users.conf
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
2008-01-09 21:37:26 +00:00
; You can define profiles for different phones specifying what files to register
; with the provisioning server. You can define either static files, or dynamically
; generated files that can have dynamic names and point to templates that variables
Remove a bunch of trailing whitespace in preparation for reformatting/cleanup. Let's try that again, this time removing trailing whitespace and not leading whitespace. I can't believe no one noticed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 14:39:21 +00:00
; can be substituted into. You can also set arbitrary variables for the profiles
res_phoneprov: Refactor phoneprov to allow pluggable config providers This patch makes res_phoneprov more modular so other modules (like pjsip) can provide configuration information instead of res_phoneprov relying solely on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API is now exposed which allows config providers to register themselves, set defaults (server profile, etc) and add user extensions. * ast_phoneprov_provider_register registers the provider and provides callbacks for loading default settings and loading users. * ast_phoneprov_provider_unregister clears the defaults and users. * ast_phoneprov_add_extension should be called once for each user/extension by the provider's load_users callback to add them. * ast_phoneprov_delete_extension deletes one extension. * ast_phoneprov_delete_extensions deletes all extensions for the provider. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3970/ ........ Merged revisions 424963 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424964 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424965 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-09 17:46:23 +00:00
; templates to have access to. Profiles are shared across all config providers.
; Example:
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
2008-01-09 21:37:26 +00:00
;[example]
;mime_type => application/octet-stream
;static_file => example/firmware
;static_file => example/default.cfg,text/xml
;${TOUPPER(${MAC})}.cfg => templates/example-mac.cfg
;setvar => DB_CIDNAME=${ODBC_CID_NAME_LOOKUP(${USERNAME})}
; Dynamically generated files have a filename registered with variable substitution
res_phoneprov: Refactor phoneprov to allow pluggable config providers This patch makes res_phoneprov more modular so other modules (like pjsip) can provide configuration information instead of res_phoneprov relying solely on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API is now exposed which allows config providers to register themselves, set defaults (server profile, etc) and add user extensions. * ast_phoneprov_provider_register registers the provider and provides callbacks for loading default settings and loading users. * ast_phoneprov_provider_unregister clears the defaults and users. * ast_phoneprov_add_extension should be called once for each user/extension by the provider's load_users callback to add them. * ast_phoneprov_delete_extension deletes one extension. * ast_phoneprov_delete_extensions deletes all extensions for the provider. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3970/ ........ Merged revisions 424963 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424964 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424965 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-09 17:46:23 +00:00
; with variables obtained from various config providers. The default provider
users.conf: Remove deprecated users.conf integration. users.conf was deprecated in Asterisk 21 and is now being removed for Asterisk 23, in accordance with the Asterisk deprecation policy. This consists of: * Removing integration with app_directory, app_voicemail, chan_dahdi, chan_iax2, and AMI. * users.conf was also partially used for res_phoneprov, and this remaining functionality is consolidated to a separate phoneprov_users.conf, used only by res_phoneprov. Resolves: #1292 UpgradeNote: users.conf has been removed and all channel drivers must be configured using their specific configuration files. The functionality previously in users.conf for res_phoneprov is now in phoneprov_users.conf.
2025-07-09 09:30:43 -04:00
; embedded in res_phoneprov reads phoneprov_users.conf. Other providers will have their own
res_phoneprov: Refactor phoneprov to allow pluggable config providers This patch makes res_phoneprov more modular so other modules (like pjsip) can provide configuration information instead of res_phoneprov relying solely on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API is now exposed which allows config providers to register themselves, set defaults (server profile, etc) and add user extensions. * ast_phoneprov_provider_register registers the provider and provides callbacks for loading default settings and loading users. * ast_phoneprov_provider_unregister clears the defaults and users. * ast_phoneprov_add_extension should be called once for each user/extension by the provider's load_users callback to add them. * ast_phoneprov_delete_extension deletes one extension. * ast_phoneprov_delete_extensions deletes all extensions for the provider. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3970/ ........ Merged revisions 424963 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424964 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424965 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-09 17:46:23 +00:00
; sources for the variables and may provide additional variables not listed here.
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
2008-01-09 21:37:26 +00:00
users.conf: Remove deprecated users.conf integration. users.conf was deprecated in Asterisk 21 and is now being removed for Asterisk 23, in accordance with the Asterisk deprecation policy. This consists of: * Removing integration with app_directory, app_voicemail, chan_dahdi, chan_iax2, and AMI. * users.conf was also partially used for res_phoneprov, and this remaining functionality is consolidated to a separate phoneprov_users.conf, used only by res_phoneprov. Resolves: #1292 UpgradeNote: users.conf has been removed and all channel drivers must be configured using their specific configuration files. The functionality previously in users.conf for res_phoneprov is now in phoneprov_users.conf.
2025-07-09 09:30:43 -04:00
; Built in variables and the options in phoneprov_users.conf that they come from
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
2008-01-09 21:37:26 +00:00
; MAC (macaddress)
; USERNAME (username)
; DISPLAY_NAME (fullname)
; SECRET (secret)
; LABEL (label)
; CALLERID (cid_number)
; VOCIEMAIL_EXTEN (vmexten)
; EXTENSION_LENGTH (localextenlength)
Merged revisions 291284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291284 | lmadsen | 2010-10-12 12:20:43 -0500 (Tue, 12 Oct 2010) | 15 lines Merged revisions 291280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r291280 | lmadsen | 2010-10-12 12:20:02 -0500 (Tue, 12 Oct 2010) | 7 lines Add undocumented variables to phoneprov.conf.sample (closes issue #18107) Reported by: lathama Patches: phoneprov.conf.sample.diff uploaded by lathama (license 1028) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-12 17:21:15 +00:00
; LINE
; LINEKEYS
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
2008-01-09 21:37:26 +00:00
; Built-in variables and the options in phoneprov.conf that they come from
; SERVER (server)
; SERVER_PORT (serverport)
Merged revisions 291284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291284 | lmadsen | 2010-10-12 12:20:43 -0500 (Tue, 12 Oct 2010) | 15 lines Merged revisions 291280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r291280 | lmadsen | 2010-10-12 12:20:02 -0500 (Tue, 12 Oct 2010) | 7 lines Add undocumented variables to phoneprov.conf.sample (closes issue #18107) Reported by: lathama Patches: phoneprov.conf.sample.diff uploaded by lathama (license 1028) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-12 17:21:15 +00:00
; Built-in variables for managing timezones and daylight savings time.
; TZOFFSET
; DST_ENABLE
; DST_START_MONTH
; DST_START_MDAY
; DST_START_HOUR
; DST_END_MONTH
; DST_END_MDAY
; DST_END_HOUR
; TIMEZONE
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
2008-01-09 21:37:26 +00:00
[polycom]
staticdir => configs/ ; Sub directory of AST_DATA_DIR/phoneprov that static files reside
Remove a bunch of trailing whitespace in preparation for reformatting/cleanup. Let's try that again, this time removing trailing whitespace and not leading whitespace. I can't believe no one noticed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 14:39:21 +00:00
; in. This allows a request to /phoneprov/sip.cfg to pull the file
; from /phoneprov/configs/sip.cfg
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
2008-01-09 21:37:26 +00:00
mime_type => text/xml ; Default mime type to use if one isn't specified or the
Remove a bunch of trailing whitespace in preparation for reformatting/cleanup. Let's try that again, this time removing trailing whitespace and not leading whitespace. I can't believe no one noticed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 14:39:21 +00:00
; extension isn't recognized
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
2008-01-09 21:37:26 +00:00
static_file => bootrom.ld,application/octet-stream ; Static files the phone will download
static_file => bootrom.ver,plain/text ; static_file => filename,mime-type
static_file => sip.ld,application/octet-stream
static_file => sip.ver,plain/text
static_file => sip.cfg
static_file => custom.cfg
Remove a bunch of trailing whitespace in preparation for reformatting/cleanup. Let's try that again, this time removing trailing whitespace and not leading whitespace. I can't believe no one noticed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 14:39:21 +00:00
static_file => 2201-06642-001.bootrom.ld,application/octet-stream
static_file => 2201-06642-001.sip.ld,application/octet-stream
static_file => 2345-11000-001.bootrom.ld,application/octet-stream
Add more polycom firmware files to static mapping git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-28 22:26:35 +00:00
static_file => 2345-11300-001.bootrom.ld,application/octet-stream
static_file => 2345-11300-010.bootrom.ld,application/octet-stream
static_file => 2345-11300-010.sip.ld,application/octet-stream
Remove a bunch of trailing whitespace in preparation for reformatting/cleanup. Let's try that again, this time removing trailing whitespace and not leading whitespace. I can't believe no one noticed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 14:39:21 +00:00
static_file => 2345-11402-001.bootrom.ld,application/octet-stream
static_file => 2345-11402-001.sip.ld,application/octet-stream
static_file => 2345-11500-001.bootrom.ld,application/octet-stream
static_file => 2345-11500-010.bootrom.ld,application/octet-stream
static_file => 2345-11500-020.bootrom.ld,application/octet-stream
static_file => 2345-11500-030.bootrom.ld,application/octet-stream
static_file => 2345-11500-030.sip.ld,application/octet-stream
static_file => 2345-11500-040.bootrom.ld,application/octet-stream
static_file => 2345-11500-040.sip.ld,application/octet-stream
static_file => 2345-11600-001.bootrom.ld,application/octet-stream
static_file => 2345-11600-001.sip.ld,application/octet-stream
static_file => 2345-11605-001.bootrom.ld,application/octet-stream
static_file => 2345-11605-001.sip.ld,application/octet-stream
static_file => 2345-12200-001.bootrom.ld,application/octet-stream
static_file => 2345-12200-001.sip.ld,application/octet-stream
static_file => 2345-12200-002.bootrom.ld,application/octet-stream
static_file => 2345-12200-002.sip.ld,application/octet-stream
Add more polycom firmware files to static mapping git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-28 22:26:35 +00:00
static_file => 2345-12200-004.bootrom.ld,application/octet-stream
static_file => 2345-12200-004.sip.ld,application/octet-stream
static_file => 2345-12200-005.bootrom.ld,application/octet-stream
static_file => 2345-12200-005.sip.ld,application/octet-stream
Merged revisions 301731 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301731 | lmadsen | 2011-01-13 11:01:43 -0600 (Thu, 13 Jan 2011) | 15 lines Merged revisions 301730 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301730 | lmadsen | 2011-01-13 11:01:11 -0600 (Thu, 13 Jan 2011) | 7 lines Add static entry for split Polycom 332 firmware. (closes issue #18607) Reported by: cjacobsen Patches: polycom_331.diff uploaded by cjacobsen (license 1029) Tested by: lathama ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-13 17:02:34 +00:00
static_file => 2345-12365-001.bootrom.ld,application/octet-stream
static_file => 2345-12365-001.sip.ld,application/octet-stream
Add more polycom firmware files to static mapping git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-28 22:26:35 +00:00
static_file => 2345-12500-001.bootrom.ld,application/octet-stream
Remove a bunch of trailing whitespace in preparation for reformatting/cleanup. Let's try that again, this time removing trailing whitespace and not leading whitespace. I can't believe no one noticed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 14:39:21 +00:00
static_file => 2345-12500-001.sip.ld,application/octet-stream
static_file => 2345-12560-001.bootrom.ld,application/octet-stream
static_file => 2345-12560-001.sip.ld,application/octet-stream
static_file => 2345-12600-001.bootrom.ld,application/octet-stream
Add more polycom firmware files to static mapping git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-28 22:26:35 +00:00
static_file => 2345-12600-001.sip.ld,application/octet-stream
static_file => 2345-12670-001.bootrom.ld,application/octet-stream
static_file => 2345-12670-001.sip.ld,application/octet-stream
static_file => 3111-15600-001.bootrom.ld,application/octet-stream
static_file => 3111-15600-001.sip.ld,application/octet-stream
static_file => 3111-40000-001.bootrom.ld,application/octet-stream
static_file => 3111-40000-001.sip.ld,application/octet-stream
Add internationalization to sample configuration file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-12 15:10:25 +00:00
static_file => SoundPointIPWelcome.wav,application/octet-stream
static_file => SoundPointIPLocalization/Japanese_Japan/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Norwegian_Norway/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Spanish_Spain/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Portuguese_Portugal/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/English_United_Kingdom/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/English_United_States/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Russian_Russia/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Italian_Italy/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Chinese_China/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Swedish_Sweden/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/English_Canada/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/German_Germany/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/French_France/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Danish_Denmark/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Dutch_Netherlands/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Korean_Korea/SoundPointIP-dictionary.xml
Add more polycom firmware files to static mapping git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-28 22:26:35 +00:00
Make phoneprov case-insensitive to remove the func_strings dependency of the default config git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147854 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09 17:04:11 +00:00
${MAC}.cfg => 000000000000.cfg ; Dynamically generated files.
${MAC}-phone.cfg => 000000000000-phone.cfg ; (relative to AST_DATA_DIR/phoneprov)
Remove a bunch of trailing whitespace in preparation for reformatting/cleanup. Let's try that again, this time removing trailing whitespace and not leading whitespace. I can't believe no one noticed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 14:39:21 +00:00
config/${MAC} => polycom.xml ; Dynamic Filename => template file
Make phoneprov case-insensitive to remove the func_strings dependency of the default config git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147854 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09 17:04:11 +00:00
${MAC}-directory.xml => 000000000000-directory.xml
Added a new module, res_phoneprov, which allows auto-provisioning of phones based on configuration templates that use Asterisk dialplan function and variable substitution. It should be possible to create phone profiles and templates that work for the majority of phones provisioned over http. It is currently only intended to provision a single user account per phone. An example profile and set of templates for Polycom phones is provided. NOTE: Polycom firmware is not included, but should be placed in AST_DATA_DIR/phoneprov/configs to match up with the included templates. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97634 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-09 21:37:26 +00:00
setvar => CUSTOM_CONFIG=/var/lib/asterisk/phoneprov/configs/custom.cfg ; Custom variable
res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support (issue #18713) Reported by: lathama Patches: snom_dir.diff uploaded by lathama (license 1028) Tested by: lathama git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03 16:13:40 +00:00
;snom 300, 320, 360, 370, 820, 821, 870 support
snom-${MAC}.xml => snom-mac.xml
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