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chan_mgcp: Remove deprecated module.

Browse Source 
Also removes res_pktcops to avoid merge conflicts
with ASTERISK~30301.

ASTERISK-30299

Change-Id: I41a316d327646a197b6f112f7f637aceb5111b41
This commit is contained in:
Mike Bradeen
2022-11-15 10:03:21 -07:00
committed by Friendly Automation
parent 841107f294
commit 89a7d30a97
21 changed files with 15 additions and 6873 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
channels/Makefile
View File

@@ -31,7 +31,6 @@ $(call MOD_ADD_C,chan_pjsip,$(wildcard pjsip/*.c))
$(call MOD_ADD_C,chan_dahdi,$(wildcard dahdi/*.c) sig_analog.c sig_pri.c sig_ss7.c)
chan_dahdi.o: _ASTCFLAGS+=$(call get_menuselect_cflags,LOTS_OF_SPANS)
chan_mgcp.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
chan_unistim.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
chan_phone.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
chan_sip.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)

5009
channels/chan_mgcp.c
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File diff suppressed because it is too large Load Diff

1
configs/samples/cli_aliases.conf.sample
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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
mgcp set debug=mgcp set debug on
abort shutdown=core abort shutdown
stop now=core stop now
stop gracefully=core stop gracefully

1
configs/samples/extensions.ael.sample
View File

@@ -423,7 +423,6 @@ context ael-default {
// };
// 6361 => Dial(IAX2/JaneDoe,,rm); // ring without time limit
// 6389 => Dial(MGCP/aaln/1@192.168.0.14);
// 6394 => Dial(Local/6275/n); // this will dial ${MARK}
// 6275 => &ael-stdexten(6275,${MARK}); // assuming ${MARK} is something like DAHDI/2

3
configs/samples/extensions.conf.sample
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@@ -223,7 +223,7 @@ TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;
; so that dialtone remains even after dialing a 9. Please note that ignorepat
; only works with channels which receive dialtone from the PBX, such as DAHDI,
; Phone, and VPB. Other channels, such as IAX2, PJSIP, SIP and MGCP, which generate
; Phone, and VPB. Other channels, such as IAX2, PJSIP, and SIP, which generate
; generate their own dialtone and converse with the PBX only after a number is
; complete, are generally unaffected by ignorepat (unless DISA or another method
; is used to generate a dialtone after answering the channel).
@@ -694,7 +694,6 @@ include => demo
;exten => 6245,s+1,Hangup ; s+1, same as n
;exten => 6245,dial+101,VoiceMail(6245,b) ; Voicemail (busy)
;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit
;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
;exten => 6391,1,Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org) ;Dial via jingle using asterisk as the transport and calling mogorman.
;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}

142
configs/samples/mgcp.conf.sample
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@@ -1,142 +0,0 @@
;
; MGCP Configuration for Asterisk
;
[general]
;port = 2427
;bindaddr = 0.0.0.0
; 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.
;cos=3 ; Sets 802.1p priority for signaling packets.
;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
; --------------------- DIGIT TIMEOUTS ----------------------------
firstdigittimeout = 30000 ; default 16000 = 16s
gendigittimeout = 10000 ; default 8000 = 8s
matchdigittimeout = 5000 ; defaults 3000 = 3s
; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; MGCP 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 MGCP channel can accept jitter,
; thus an enabled jitterbuffer on the receive MGCP side will only
; be used if the sending side can create jitter and jbforce is
; also set to yes.
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a MGCP
; 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 MGCP
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
; The option represents the number of milliseconds by which the new
; jitter buffer will pad its size. the default is 40, so without
; modification, the new jitter buffer will set its size to the jitter
; value plus 40 milliseconds. increasing this value may help if your
; network normally has low jitter, but occasionally has spikes.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
; ----------------------------------------------------------------------------------
;[dlinkgw]
;host = 192.168.0.64
;context = default
;directmedia = no
;line => aaln/2
;line => aaln/1
;; The MGCP channel supports the following service codes:
;; # - Transfer
;; *67 - Calling Number Delivery Blocking
;; *70 - Cancel Call Waiting
;; *72 - Call Forwarding Activation
;; *73 - Call Forwarding Deactivation
;; *78 - Do Not Disturb Activation
;; *79 - Do Not Disturb Deactivation
;; *8 - Call pick-up
;
; known to work with Swissvoice IP10s
;[192.168.1.20]
;context=local
;host=192.168.1.20
;callerid = "John Doe" <123>
;callgroup=0 ; in the range from 0 to 63
;pickupgroup=0 ; in the range from 0 to 63
;nat=no
;threewaycalling=yes
;transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer
;callwaiting=yes ; this might be a cause of trouble for ip10s
;cancallforward=yes
;line => aaln/1
;
;[dph100]
;
; Supporting the DPH100M requires defining DLINK_BUGGY_FIRMWARE in
; chan_mgcp.c in addition to enabling the slowsequence mode due to
; bugs in the D-Link firmware
;
;context=local
;host=dynamic
;dtmfmode=none ; DTMF Mode can be 'none', 'rfc2833', or 'inband' or
; 'hybrid' which starts in none and moves to inband. Default is none.
;slowsequence=yes ; The DPH100M does not follow MGCP standards for sequencing
;line => aaln/1
; known to work with wave7optics FTTH LMGs
;[192.168.1.20]
;accountcode = 1000 ; record this in cdr as account identification for billing
;amaflags = billing ; record this in cdr as flagged for 'billing',
; 'documentation', or 'omit'
;context = local
;host = 192.168.1.20
;wcardep = aaln/* ; enables wildcard endpoint and sets it to 'aaln/*'
; another common format is '*'
;callerid = "Duane Cox" <123> ; now lets setup line 1 using per endpoint configuration...
;callwaiting = no
;callreturn = yes
;cancallforward = yes
;directmedia = no
;transfer = no
;dtmfmode = inband
;setvar=one=1 ; Set channel variables associated with this incoming line
;setvar=two=2
;line => aaln/1 ; now lets save this config to line1 aka aaln/1
;clearvars=all ; Reset list of variables back to none
;callerid = "Duane Cox" <456> ; now lets setup line 2
;callwaiting = no
;callreturn = yes
;cancallforward = yes
;directmedia = no
;transfer = no
;dtmfmode = inband
;line => aaln/2 ; now lets save this config to line2 aka aaln/2
; PacketCable
;[sbv5121e-mta.test.local]
;host = 10.0.1.3
;callwaiting = 1
;canreinvite = 1
;dtmfmode = rfc2833
;amaflags = BILLING
;ncs = yes ; Use NCS 1.0 signalling
;pktcgatealloc = yes ; Allocate DQOS gate on CMTS
;hangupongateremove = yes ; Hangup the channel if the CMTS close the gate
;callerid = 3622622225
;accountcode = test-3622622225
;line = aaln/1
;callerid = 3622622226
;accountcode = test-3622622226
;line = aaln/2

32
configs/samples/res_pktccops.conf.sample
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@@ -1,32 +0,0 @@
;; Sample res_pktccops.conf
;
;[general]
;gateinfoperiod => 60 ; default 60s
;gatetimeout = 150 ; default 150
;t1 => 250 ; default 250s
;t7 => 200 ; default 200s
;t8 => 300 ; default 300s
;keepalive => 60 ; default 60s
;
;[teszt]
;host => 192.168.0.24
;pool => 10.0.1.0 10.0.1.255
;pool => 10.0.3.0 10.0.3.255
;pool => 10.0.7.0 10.0.8.255
;pool => 10.0.10.0 10.0.11.255
;
;[general]
;gateinfoperiod => 60 ; default 60s
;gatetimeout = 150 ; default 150
;t1 => 250 ; default 250s
;t7 => 200 ; default 200s
;t8 => 300 ; default 300s
;keepalive => 60 ; default 60s
;
;[test]
;host => 192.168.0.24
;pool => 10.0.1.0 10.0.1.255
;pool => 10.0.3.0 10.0.3.255
;pool => 10.0.7.0 10.0.8.255
;pool => 10.0.10.0 10.0.11.255
;

95
configure vendored
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@@ -34779,101 +34779,6 @@ printf "%s\n" "#define HAVE_LINUX_COMPILER_H 1" >>confdefs.h
fi
# Used in res/res_pktccops
if test "x${PBX_MSG_NOSIGNAL}" != "x1"; then
{ printf "%s\n" "$as_me:${as_lineno-$LINENO}: checking for MSG_NOSIGNAL in sys/socket.h" >&5
printf %s "checking for MSG_NOSIGNAL in sys/socket.h... " >&6; }
saved_cppflags="${CPPFLAGS}"
if test "x${MSG_NOSIGNAL_DIR}" != "x"; then
MSG_NOSIGNAL_INCLUDE="-I${MSG_NOSIGNAL_DIR}/include"
fi
CPPFLAGS="${CPPFLAGS} ${MSG_NOSIGNAL_INCLUDE}"
cat confdefs.h - <<_ACEOF >conftest.$ac_ext
/* end confdefs.h. */
#include <sys/socket.h>
int
main (void)
{
#if defined(MSG_NOSIGNAL)
int foo = 0;
#else
int foo = bar;
#endif
0
;
return 0;
}
_ACEOF
if ac_fn_c_try_compile "$LINENO"
then :
{ printf "%s\n" "$as_me:${as_lineno-$LINENO}: result: yes" >&5
printf "%s\n" "yes" >&6; }
PBX_MSG_NOSIGNAL=1
printf "%s\n" "#define HAVE_MSG_NOSIGNAL 1" >>confdefs.h
else $as_nop
{ printf "%s\n" "$as_me:${as_lineno-$LINENO}: result: no" >&5
printf "%s\n" "no" >&6; }
fi
rm -f core conftest.err conftest.$ac_objext conftest.beam conftest.$ac_ext
CPPFLAGS="${saved_cppflags}"
fi
if test "x${PBX_SO_NOSIGPIPE}" != "x1"; then
{ printf "%s\n" "$as_me:${as_lineno-$LINENO}: checking for SO_NOSIGPIPE in sys/socket.h" >&5
printf %s "checking for SO_NOSIGPIPE in sys/socket.h... " >&6; }
saved_cppflags="${CPPFLAGS}"
if test "x${SO_NOSIGPIPE_DIR}" != "x"; then
SO_NOSIGPIPE_INCLUDE="-I${SO_NOSIGPIPE_DIR}/include"
fi
CPPFLAGS="${CPPFLAGS} ${SO_NOSIGPIPE_INCLUDE}"
cat confdefs.h - <<_ACEOF >conftest.$ac_ext
/* end confdefs.h. */
#include <sys/socket.h>
int
main (void)
{
#if defined(SO_NOSIGPIPE)
int foo = 0;
#else
int foo = bar;
#endif
0
;
return 0;
}
_ACEOF
if ac_fn_c_try_compile "$LINENO"
then :
{ printf "%s\n" "$as_me:${as_lineno-$LINENO}: result: yes" >&5
printf "%s\n" "yes" >&6; }
PBX_SO_NOSIGPIPE=1
printf "%s\n" "#define HAVE_SO_NOSIGPIPE 1" >>confdefs.h
else $as_nop
{ printf "%s\n" "$as_me:${as_lineno-$LINENO}: result: no" >&5
printf "%s\n" "no" >&6; }
fi
rm -f core conftest.err conftest.$ac_objext conftest.beam conftest.$ac_ext
CPPFLAGS="${saved_cppflags}"
fi
if test "x${PBX_SDL}" != "x1" -a "${USE_SDL}" != "no"; then

3
configure.ac
View File

@@ -2786,9 +2786,6 @@ fi
AC_CHECK_HEADER([linux/compiler.h],
[AC_DEFINE_UNQUOTED([HAVE_LINUX_COMPILER_H], 1, [Define to 1 if your system has linux/compiler.h.])])
# Used in res/res_pktccops
AST_C_DEFINE_CHECK([MSG_NOSIGNAL], [MSG_NOSIGNAL], [sys/socket.h])
AST_C_DEFINE_CHECK([SO_NOSIGPIPE], [SO_NOSIGPIPE], [sys/socket.h])
AST_EXT_TOOL_CHECK([SDL], [sdl-config])
AST_EXT_LIB_CHECK([SDL_IMAGE], [SDL_image], [IMG_Load], [SDL_image.h], [${SDL_LIB}], [${SDL_INCLUDE}])

7
doc/UPGRADE-staging/chan_mgcp_removal.txt Normal file
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@@ -0,0 +1,7 @@
Subject: chan_mgcp
Master-Only: True
This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.

4
doc/asterisk.8
View File

@@ -45,8 +45,8 @@ 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,
H.323, Unistim) devices (both endpoints and proxies), as well as traditional TDM
range of telephony devices including packet voice (SIP, IAX2, 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.
.PP

4
doc/asterisk.sgml
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@@ -52,8 +52,8 @@
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,
H.323, Unistim) devices (both endpoints and proxies), as well as traditional TDM
range of telephony devices including packet voice (SIP, IAX2 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.
</para>

2
include/asterisk/doxyref.h
View File

@@ -250,7 +250,7 @@
/*!
* \page Config_rtp RTP configuration
* \arg Implemented in \ref rtp.c
* Used in \ref chan_sip.c and \ref chan_mgcp.c (and various H.323 channels)
* Used in \ref chan_sip.c (and various H.323 channels)
* \section rtpconf rtp.conf
* \verbinclude rtp.conf.sample
*/

82
include/asterisk/pktccops.h
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@@ -1,82 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2009, Attila Domjan
*
* Attila Domjan <attila.domjan.hu@gmail.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief PacketCable COPS
*
* \author Attila Domjan <attila.domjan.hu@gmail.com>
*/
#ifndef _ASTERISK_PKTCCOPS_H
#define _ASTERISK_PKTCCOPS_H
#include "asterisk/optional_api.h"
#if defined(__cplusplus) || defined(c_plusplus)
extern "C" {
#endif
enum {
GATE_SET,
GATE_INFO,
GATE_SET_HAVE_GATEID,
GATE_DEL
};
enum {
GATE_ALLOC_FAILED,
GATE_ALLOC_PROGRESS,
GATE_ALLOCATED,
GATE_CLOSED,
GATE_CLOSED_ERR,
GATE_OPEN,
GATE_DELETED,
GATE_TIMEOUT
};
struct cops_gate {
AST_LIST_ENTRY(cops_gate) list;
uint32_t gateid;
uint16_t trid;
time_t in_transaction;
uint32_t mta;
int state;
time_t allocated;
time_t checked;
time_t deltimer;
struct cops_cmts *cmts;
int (* got_dq_gi) (struct cops_gate *gate);
int (* gate_remove) (struct cops_gate *gate);
int (* gate_open) (struct cops_gate *gate);
void *tech_pvt;
};
AST_OPTIONAL_API(struct cops_gate *, ast_pktccops_gate_alloc,
(int cmd, struct cops_gate *gate, uint32_t mta, uint32_t actcount,
float bitrate, uint32_t psize, uint32_t ssip, uint16_t ssport,
int (* const got_dq_gi) (struct cops_gate *gate),
int (* const gate_remove) (struct cops_gate *gate)),
{ return NULL; });
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif
#endif /* _ASTERISK_PKTCCOPS_H */

1
include/asterisk/res_hep.h
View File

@@ -42,7 +42,6 @@ enum hepv3_capture_type {
HEPV3_CAPTURE_TYPE_SDP = 0x03,
HEPV3_CAPTURE_TYPE_RTP = 0x04,
HEPV3_CAPTURE_TYPE_RTCP = 0x05,
HEPV3_CAPTURE_TYPE_MGCP = 0x06,
HEPV3_CAPTURE_TYPE_MEGACO = 0x07,
HEPV3_CAPTURE_TYPE_M2UA = 0x08,
HEPV3_CAPTURE_TYPE_M3UA = 0x09,

1
main/asterisk.dynamics
View File

@@ -2,7 +2,6 @@
*ast_adsi_*;
*ast_agi_*;
*ast_beep_*;
*ast_pktccops_*;
*ast_smdi_*;
*ast_monitor_*;
*ast_key_get;

2
menuselect/example_menuselect-tree
View File

@@ -198,8 +198,6 @@
</member>
<member name="chan_local" displayname="Local Proxy Channel" remove_on_change="channels/chan_local.o channels/chan_local.so">
</member>
<member name="chan_mgcp" displayname="Media Gateway Control Protocol (MGCP)" remove_on_change="channels/chan_mgcp.o channels/chan_mgcp.so">
</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_zap" displayname="Zapata Telephony" remove_on_change="channels/chan_zap.o channels/chan_zap.so">

2
menuselect/test/menuselect-tree
View File

@@ -228,8 +228,6 @@
</member>
<member name="chan_local" displayname="Local Proxy Channel (Note: used internally by other modules)" remove_on_change="channels/chan_local.o channels/chan_local.so">
</member>
<member name="chan_mgcp" displayname="Media Gateway Control Protocol (MGCP)" remove_on_change="channels/chan_mgcp.o channels/chan_mgcp.so">
</member>
<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>

1486
res/res_pktccops.c
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File diff suppressed because it is too large Load Diff

6
res/res_pktccops.exports.in
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@@ -1,6 +0,0 @@
{
global:
LINKER_SYMBOL_PREFIXast_pktccops_gate_alloc;
local:
*;
};

4
tests/CI/buildAsterisk.sh
View File

@@ -156,13 +156,13 @@ 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_unistim"
mod_disables+=" chan_alsa chan_console 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"
mod_disables+=" pbx_dundi pbx_loopback"
mod_disables+=" res_ael_share res_calendar res_config_ldap res_config_pgsql res_corosync"
mod_disables+=" res_http_post res_pktccops res_rtp_multicast res_snmp res_xmpp"
mod_disables+=" res_http_post res_rtp_multicast res_snmp res_xmpp"
fi
runner menuselect/menuselect `gen_mods disable $mod_disables` menuselect.makeopts