chan_skinny: Remove deprecated module.

ASTERISK-30300

Change-Id: I8be11455010b8ec552e62b0719368342e8a1bae9
This commit is contained in:
Mike Bradeen 2022-11-16 08:51:25 -07:00 committed by Friendly Automation
parent 1c8acdb1c0
commit 8d652ab4be
10 changed files with 9 additions and 9072 deletions

File diff suppressed because it is too large Load Diff

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@ -174,7 +174,6 @@ agi dumphtml=agi dump html
ael debug=ael set debug
funcdevstate list=devstate list
sip history=sip set history on
skinny debug=skinny set debug on
mgcp set debug=mgcp set debug on
abort shutdown=core abort shutdown
stop now=core stop now

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@ -1,207 +0,0 @@
;
; Skinny Configuration for Asterisk
;
[general]
bindaddr=0.0.0.0 ; Address to bind to
bindport=2000 ; Port to bind to, default tcp/2000
dateformat=M-D-Y ; M,D,Y in any order (6 chars max)
; "A" may also be used, but it must be at the end.
; Use M for month, D for day, Y for year, A for 12-hour time.
keepalive=120
;authtimeout = 30 ; authtimeout specifies the maximum number of seconds a
; client has to authenticate. If the client does not
; authenticate beofre this timeout expires, the client
; will be disconnected. (default: 30 seconds)
;authlimit = 50 ; authlimit specifies the maximum number of
; unauthenticated sessions that will be allowed to
; connect at any given time. (default: 50)
;vmexten=8500 ; Systemwide voicemailmain pilot number
; It must be in the same context as the calling
; device/line
; If regcontext is specified, Asterisk will dynamically create and destroy a
; NoOp priority 1 extension for a given line which registers or unregisters with
; us and have a "regexten=" configuration item.
; Multiple contexts may be specified by separating them with '&'. The
; actual extension is the 'regexten' parameter of the registering line or its
; name if 'regexten' is not provided. If more than one context is provided,
; the context must be specified within regexten by appending the desired
; context after '@'. More than one regexten may be supplied if they are
; separated by '&'. Patterns may be used in regexten.
;
;regcontext=skinnyregistrations
;allow=all ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
; for framing options
;disallow=
; The imeddialkey option allows for a key to be used to immediately dial the already
; entered number. This is useful where the dialplan includes variable length pattern
; matching. Valid options are '#' and '*'. On devices with soft buttons, a button will
; be available to immediately dial when a pattern than can be dialed has been entered.
; Default is unset, that is no immediated dial key (softbutton still exists).
;
;immeddialkey=#
; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
;tos=cs3 ; Sets TOS for signaling packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;cos=3 ; Sets 802.1p priority for signaling packets.
;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
;cos_video=4 ; Sets 802.1p priority for RTP video packets.
; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
;jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; skinny channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The skinny channel can accept
; jitter, thus a jitterbuffer on the receive skinny side will be
; used only if it is forced and enabled.
;jbforce = no ; Forces the use of a jitterbuffer on the receive side of a skinny
; channel. Defaults to "no".
;jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
;jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
;jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a
; skinny channel. Two implementations are currently available
; - "fixed" (with size always equals to jbmaxsize)
; - "adaptive" (with variable size, actually the new jb of IAX2).
; Defaults to fixed.
;jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
; ----------------------------------------------------------------------------------
[lines]
; ---------------------------------- LINES SECTION --------------------------------
; Options set under [lines] apply to all lines unless explicitly set for a particular
; device. The options that can be set under lines are specified in GENERAL LINE OPTIONS.
; These options can also be set for each individual device as well as those under SPECIFIC
; LINE OPTIONS.
;
; Each label below [lines] in [] is a new line with the specific options specified below
; it. Config stops reading new lines when one of the following is found: [general], [devices]
; or the end of skinny.conf.
;
; Where options are common to both lines and devices, the results typically take that of
; the least permission. ie if a no is set for either line or device, the call will not be
; able to use that permission
; ------------------------------- GENERAL LINE OPTIONS -----------------------------
;earlyrtp=1 ; whether audio signalling should be provided by asterisk
; ; (earlyrtp=1) or device generated (earlyrtp=0). default=yes
;transfer=1 ; whether the device is allowed to transfer. default=yes
;context=default ; context to use for this line.
;callfwdtimeout=20000 ; ms before cfwd_noans occurs (default 20 secs)
; ------------------------------ SPECIFIC LINE OPTIONS -----------------------------
;setvar= ; allows for the setting of chanvars.
; ----------------------------------------------------------------------------------
;[100]
;nat=yes
;callerid="Customer Support" <810-234-1212>
; Note: app_voicemail mailboxes must be in the form of mailbox@context.
;mailbox=100
;vmexten=8500 ; Device level voicemailmain pilot number
;regexten=100
;context=inbound
;linelabel="Support Line" ; Displays next to the line
; button on 7940's and 7960s
;[110]
;callerid="John Chambers" <408-526-4000>
;context=did
;regexten=110
;linelabel="John"
;mailbox=110
;[120]
;Nothing set, so all the defaults are used
;[500]
;nat=yes
;callerid="George W. Bush" <202-456-1414>
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
; cause the given audio file to
; be played upon completion of
; an attended transfer to the
; target of the transfer.
;mailbox=500
;callwaiting=yes
;transfer=yes
;threewaycalling=yes
;context=default
;mohinterpret=default ; This option specifies a default music on hold class to
; use when put on hold if the channel's moh class was not
; explicitly set with Set(CHANNEL(musicclass)=whatever) and
; the peer channel did not suggest a class to use.
;mohsuggest=default ; This option specifies which music on hold class to suggest to the peer channel
; when this channel places the peer on hold. It may be specified globally or on
; a per-user or per-peer basis.
[devices]
; --------------------------------- DEVICES SECTION -------------------------------
; Options set under [devices] apply to all devices unless explicitly set for a particular
; device. The options that can be set under devices are specified in GENERAL DEVICE OPTIONS.
; These options can also be set for each individual device as well as those under SPECIFIC
; DEVICE OPTIONS.
;
; Each label below [devices] in [] is a new device with the specific options specified below
; it. Config stop reading new devices when one of the following is found: [general], [lines]
; or the end of skinny.conf.
;
; Where options are common to both lines and devices, the results typically take that of
; the least permission. ie if a no is set for either line or device, the call will not be
; able to use that permission
; ------------------------------ GENERAL DEVICE OPTIONS ----------------------------
;earlyrtp=1 ; whether audio signalling should be provided by asterisk
; ; (earlyrtp=1) or device generated (earlyrtp=0). default=yes
;transfer=1 ; whether the device is allowed to transfer. default=yes
; ----------------------------- SPECIFIC DEVICE OPTIONS ----------------------------
;device="SEPxxxxxxxxxxxx ; id of the device. Must be set.
;version=P002G204 ; firmware version to be loaded. If this version is different
; ; to the one on the device, the device will try to load this
; ; version from the tftp server. Set to device firmware version.
; ----------------------------------------------------------------------------------
; Typical config for 12SP+
;[florian]
;device=SEP00D0BA847E6B
;version=P002G204 ; Thanks critch
;context=did
;directmedia=yes ; Allow media to go directly between two RTP endpoints.
;line=120 ; Dial(Skinny/120@florian)
; Service URLs attached to line buttons (eg phone directory)
; See http://www.voip-info.org/wiki/view/Asterisk+Cisco+79XX+XML+Services
; for intro to xml structure.
;serviceurl=Directory,http://host/file.xml
; Typical config for a 7910
;[duba] ; Device name
;device=SEP0007EB463101 ; Official identifier
;version=P002F202 ; Firmware version identifier
;host=192.168.1.144
;permit=192.168.0/24 ; Optional, used for authentication
;line=500
; Typical config for a 7940 with dual 7914s
;[support]
;device=SEP0007EB463121
;line=100
;line=110
;speeddial => 111,Jack Smith ; Adds a speeddial button to a device.
;speeddial => 112@hints,Bob Peterson ; When a context is specified, the speeddial watches a dialplan hint.
;addon => 7914
;addon => 7914

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@ -0,0 +1,6 @@
Subject: chan_skinny
Master-Only: True
This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.

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@ -45,7 +45,7 @@ asterisk
provides Private Branch eXchange (PBX), Interactive Voice Response (IVR),
Automated Call Distribution (ACD), Voice over IP (VoIP) gatewaying,
Conferencing, and a plethora of other telephony applications to a broad
range of telephony devices including packet voice (SIP, IAX2, MGCP, Skinny,
range of telephony devices including packet voice (SIP, IAX2, MGCP,
H.323, Unistim) devices (both endpoints and proxies), as well as traditional TDM
hardware including T1, E1, ISDN PRI, GR-303, RBS, Loopstart, Groundstart,
ISDN BRI and many more.

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@ -52,7 +52,7 @@
provides Private Branch eXchange (PBX), Interactive Voice Response (IVR),
Automated Call Distribution (ACD), Voice over IP (VoIP) gatewaying,
Conferencing, and a plethora of other telephony applications to a broad
range of telephony devices including packet voice (SIP, IAX2, MGCP, Skinny,
range of telephony devices including packet voice (SIP, IAX2, MGCP,
H.323, Unistim) devices (both endpoints and proxies), as well as traditional TDM
hardware including T1, E1, ISDN PRI, GR-303, RBS, Loopstart, Groundstart,
ISDN BRI and many more.

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@ -224,9 +224,6 @@ int ast_parking_is_exten_park(const char *context, const char *exten);
* is very difficult to do with the new asynchronous design. That information may
* not be available anywhere by the time this function currently returns.
*
* Only, chan_skinny is known to call this function and use the exten parameter
* for the phone display.
*
* \retval 0 on success
* \retval non-zero on error
*/

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@ -206,8 +206,6 @@
</member>
<member name="chan_sip" displayname="Session Initiation Protocol (SIP)" remove_on_change="channels/chan_sip.o channels/chan_sip.so">
</member>
<member name="chan_skinny" displayname="Skinny Client Control Protocol (Skinny)" remove_on_change="channels/chan_skinny.o channels/chan_skinny.so">
</member>
<member name="chan_zap" displayname="Zapata Telephony" remove_on_change="channels/chan_zap.o channels/chan_zap.so">
<depend>zaptel</depend>
</member>

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@ -237,8 +237,6 @@
<member name="chan_sip" displayname="Session Initiation Protocol (SIP)" remove_on_change="channels/chan_sip.o channels/chan_sip.so">
<depend>chan_local</depend>
</member>
<member name="chan_skinny" displayname="Skinny Client Control Protocol (Skinny)" remove_on_change="channels/chan_skinny.o channels/chan_skinny.so">
</member>
<member name="chan_unistim" displayname="UNISTIM Protocol (USTM)" remove_on_change="channels/chan_unistim.o channels/chan_unistim.so">
</member>
<member name="chan_usbradio" displayname="usb Console Channel Driver" remove_on_change="channels/chan_usbradio.o channels/chan_usbradio.so">

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@ -156,7 +156,7 @@ if [ $NO_MENUSELECT -eq 0 ] ; then
mod_disables+=" cdr_adaptive_odbc cdr_custom cdr_manager cdr_odbc cdr_pgsql cdr_radius"
mod_disables+=" cdr_tds"
mod_disables+=" cel_odbc cel_pgsql cel_radius cel_sqlite3_custom cel_tds"
mod_disables+=" chan_alsa chan_console chan_mgcp chan_motif chan_rtp chan_skinny chan_unistim"
mod_disables+=" chan_alsa chan_console chan_mgcp chan_motif chan_rtp chan_unistim"
mod_disables+=" func_frame_trace func_pitchshift func_speex func_volume func_dialgroup"
mod_disables+=" func_periodic_hook func_sprintf func_enum func_extstate func_sysinfo func_iconv"
mod_disables+=" func_callcompletion func_version func_rand func_sha1 func_module func_md5"