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chan_mgcp: Remove deprecated module.

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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
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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
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@@ -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
;