Add SIP station 24 for telegott line

This reverts commit a6b9caad5a.

.gitignore

@@ -1 +0,0 @@
-*.swp

asound.state

@@ -273,7 +273,7 @@ state.SB {
 	control.16 {
 		iface MIXER
 		name 'Master Playback Volume'
-		value 57
+		value 64
 		comment {
 			access 'read write'
 			type INTEGER
@@ -281,7 +281,7 @@ state.SB {
 			range '0 - 64'
 			dbmin -6400
 			dbmax 0
-			dbvalue.0 -700
+			dbvalue.0 0
 		}
 	}
 	control.17 {

clear-queue-members.sh

@@ -1,3 +0,0 @@
-#!/bin/sh
-
-asterisk -x 'queue show auction' | grep wait: | awk '{ print $2; }' | xargs -I ID asterisk -x 'channel request hangup ID'

extensions.conf

@@ -732,21 +732,15 @@ exten => _X.,n,Return()
 [from-eventphone]
 exten => s,1,Answer()
 same  => n,Wait(1)
+same  => n,Playback(/etc/asterisk/media/playback/e_1_herzlich_willkommen_beim_teleshopping)
 same  => n,GotoIf($[ "${DB(stream/open)}" != "1" ]?stream-closed,1)
-same  => n,Background(/etc/asterisk/media/playback/e_1_herzlich_willkommen_beim_teleshopping,m)
 same  => n(start),Background(/etc/asterisk/media/playback/e_3_dr_cken_Sie,m)
-same  => n(start),Background(/etc/asterisk/media/playback/telefon_drueckenSieDie7,m)
 same  => n,WaitExten(4)
 same  => n,Goto(start)
 exten  => 1,1,Goto(record-bid,1,1)
 exten  => 2,1,Goto(take-question,s,1)
 
-exten => 7,1,Wait(0.5)
-same => n,Queue(gott,t)
-same => n,Hangup()
-
 exten => stream-closed,1,Wait(0.8)
-same  => n,Playback(/etc/asterisk/media/playback/e_1_herzlich_willkommen_beim_teleshopping)
 same  => n,Playback(/etc/asterisk/media/playback/e_2_bei_uns_ist_leider_gerade_nichts_los)
 same  => n,MusicOnHold(shop,10)
 same  => n,Goto(1)
@@ -768,15 +762,14 @@ same  => n,GotoIf($[ "${input}" != "1" ]?start)
 same  => n,Set(FILE(/tmp/money.txt,,,al,u)=${money} ${CALLERID(num)})
 same  => n,Playback(/etc/asterisk/media/playback/e_12_vielen_dank_f_r_dein_gebot)
 same  => n,Set(QUEUE_PRIO=${money})
-;same  => n,Originate(Console/dsp,exten,auction-spy,s,1,,a)
+same  => n,Originate(Console/dsp,exten,auction-spy,s,1,,a)
 same  => n,Set(SPYGROUP=10022)
 same  => n,GotoIf($[ "${quality_assured}" != "" ]?direct_queue)
 same  => n,Playback(/etc/asterisk/media/playback/e_4_zur_qualit_tssicherung)
 same  => n,Set(quality_assured="most_definitely")
-same  => n(direct_queue),GotoIf($[ "${DB(auction/open)}" != "1" ]?bidding-closed,1)
+same  => n(direct_queue),Set(__AUCTION_BID=${money})
-same  => n,Set(__AUCTION_BID=${money})
 same  => n,Queue(auction,tc)
-same  => n,Dial(IAX2/stubnitz/21,60,U(bidding-announce))
+same  => n,Dial(SIP/21,60,U(bidding-announce))
 same  => n,Hangup()
 
 exten => bidding-closed,1,Playback(/etc/asterisk/media/playback/e_5_leider_kann_gerade_kein_gebot)
@@ -788,41 +781,32 @@ same  => n,Goto(1,1)
 
 [auction-spy]
 exten => s,1,Noop()
-same  => n,ChanSpy(SIP/22,bq)
+same  => n,ChanSpy(SIP/22,bqv(2))
 
 [bidding-announce]
 exten => s,1,Wait(1)
-;same  => n,Playback(/etc/asterisk/media/playback/e_20.1_das_gebot_der_anrufenden_dramatischer)
+same  => n,Playback(/etc/asterisk/media/playback/e_20.1_das_gebot_der_anrufenden_dramatischer)
-same  => n,Playback(/etc/asterisk/media/playback/e_20_das_gebot_der_anrufenden_betr_gt)
 same  => n,SayNumber(${AUCTION_BID})
-same  => n,Playback(/etc/asterisk/media/playback/e_7.2_eur_s)
 same  => n,Return()
 
 [bidding-queue-member]
 exten => _2X,1,Ringing()
-same  => n,Dial(IAX2/stubnitz/${EXTEN},,U(bidding-announce))
+same  => n,Dial(SIP/${EXTEN},,U(bidding-announce))
-same  => n,RemoveQueueMember(auction,Local/${EXTEN}@bidding-queue-member)
-same  => n,Hangup
 
 [take-question]
-exten => s,1,Background(/etc/asterisk/media/playback/e_4_zur_qualit_tssicherung)
+exten => s,1,Playback(/etc/asterisk/media/playback/e_4_zur_qualit_tssicherung)
 same  => n,Wait(0.5)
 same  => n,Queue(shop,t)
 same  => n,Hangup()
 
-exten => 82,1,Wait(0.5)
-same  => n,Playback(/etc/asterisk/media/playback/ata)
-same  => n,Queue(shop,t)
-same  => n,Hangup()
-
-exten => i,1,Wait(0.5)
-same  => n,Queue(shop,t)
-same  => n,Hangup()
-
 [local]
+exten => _22X,1,Dial(SIP/${EXTEN:1})
+same  => n,Hangup
 
-[fromstubnitz]
+exten => _X.,1,Dial(IAX2/hetzner/${EXTEN})
-exten => _22X,1,Dial(IAX2/stubnitz/${EXTEN:1})
+same  => n,Hangup
+
+[oldlocal]
 exten => _0X.,1,Goto(to-eventphone,${EXTEN:1},1)
 
 ; Open bidding, read start bid
@@ -863,13 +847,11 @@ same  => n,Set(DB(stream/open)=0)
 same  => n,Playback(/etc/asterisk/media/playback/e_15.2_herzlichen_gl_ckwunsch_pott)
 same  => n,Hangup()
 
-; Remove queue member
-exten => 95,1,Answer()
-same  => n,Wait(0.5)
-same  => n,RemoveQueueMember(auction,Local/22@bidding-queue-member)
-same  => n,Playback(/etc/asterisk/media/playback/e_15.2_herzlichen_gl_ckwunsch_pott)
-same  => n,Hangup()
-
 [to-eventphone]
 exten => _X.,1,SET(CALLERID(all)="Teleshop" <7467>)
 same  => n,Dial(SIP/${EXTEN}@eventphone)
+
+[fromhetzner]
+exten => _2X,1,Originate(Console/dsp,exten,auction-spy,s,1,,a)
+same  => n,Dial(SIP/${EXTEN})
+same  => n,Hangup()

http.conf

@@ -26,13 +26,13 @@ servername=Asterisk
 ; Whether HTTP/HTTPS interface is enabled or not.  Default is no.
 ; This also affects manager/rawman/mxml access (see manager.conf)
 ;
-enabled=yes
+;enabled=yes
 ;
 ; Address to bind to, both for HTTP and HTTPS. You MUST specify
 ; a bindaddr in order for the HTTP server to run. There is no
 ; default value.
 ;
-bindaddr=172.19.27.2
+;bindaddr=0.0.0.0
 ;
 ; Port to bind to for HTTP sessions (default is 8088)
 ;

iax.conf

@@ -26,7 +26,7 @@
                          ; IPv6 the address needs to be in brackets then colon
                          ; and port (e.g. bindaddr=[2001:db8::1]:4569).
 
-bindaddr=172.19.27.2    ; You can specify 'bindaddr' more than once to bind to
+bindaddr=172.19.27.1    ; You can specify 'bindaddr' more than once to bind to
                          ; multiple addresses, but the first will be the
                          ; default. IPv6 addresses are accepted.
 
@@ -149,12 +149,11 @@ bandwidth=low
 ; You can also fine tune codecs here using "allow" and "disallow" clauses with
 ; specific codecs.  Use "all" to represent all formats.
 ;
-;allow=all
+disallow=all
+allow=gsm
 ;disallow=g723.1
 ;disallow=lpc10
 ;allow=gsm
-disallow=all
-allow=gsm
 ;
 
 ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
@@ -482,9 +481,9 @@ autokill=yes
 ;
 ;shrinkcallerid=yes     ; on by default
 
-; Guest sections for unauthenticated connection attempts.  Just specify an
+;; Guest sections for unauthenticated connection attempts.  Just specify an
-; empty secret, or provide no secret section.
+;; empty secret, or provide no secret section.
-;
+;;
 ;[guest]
 ;type=user
 ;context=public
@@ -670,12 +669,12 @@ autokill=yes
 ;context=number-please ; we start at the s extension in this context
 ;
 
-[stubnitz]
+[hetzner]
 type=friend
-host=172.19.27.1
+host=172.19.27.2
-username=hetzner
+username=stubnitz
 secret=f9DuF9Sh0GRKUFrF
-context=fromstubnitz
+context=fromhetzner
 qualify=yes
 qualifyfreqok=10000
 qualifyfreqnotok=10000

iaxprov.conf

@@ -0,0 +1,80 @@
+;
+; IAX2 Provisioning Information
+;
+; Contains provisioning information for templates and for specific service
+; entries.
+;
+; Templates provide a group of settings from which provisioning takes place.
+; A template may be based upon any template that has been specified before
+; it.  If the template that an entry is based on is not specified then it is
+; presumed to be 'default' (unless it is the first of course).
+;
+; Templates which begin with 'si-' are used for provisioning units with
+; specific service identifiers.  For example the entry "si-000364000126"
+; would be used when the device with the corresponding service identifier of
+; "000364000126" attempts to register or make a call.
+;
+[default]
+;
+; The port number the device should use to bind to.  The default is 4569.
+;
+;port=4569
+;
+; server is our PRIMARY server for registration and placing calls
+;
+;server=192.168.69.3
+;
+; altserver is the BACKUP server for registration and placing calls in the
+; event the primary server is unavailable.
+;
+;altserver=192.168.69.4
+;
+; port is the port number to use for IAX2 outbound.  The connections to the
+; server and altserver -- default is of course 4569.
+;serverport=4569
+;
+; language is the preferred language for the device
+;
+;language=en
+;
+; codec is the requested codec.  The iaxy supports ulaw and adpcm
+;
+codec=ulaw
+;
+; flags is a comma separated list of flags which the device should
+; use and may contain any of the following keywords:
+;
+;  "register"  - Register with server
+;  "secure"    - Do not accept calls / provisioning not originated by the server
+;  "heartbeat" - Generate status packets on port 9999 sent to 255.255.255.255
+;  "debug"     - Output extra debugging to port 9999
+;
+;  Note that use can use += and -= to adjust parameters
+;
+flags=register,heartbeat
+;
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of this parameter.
+;tos=ef
+;
+; Example iaxy provisioning
+;
+;[si-000364000126]
+;user=iaxy
+;pass=bitsy
+;flags += debug
+
+;[si-000364000127]
+;user=iaxy2
+;pass=bitsy2
+;template=si-000364000126
+;flags += debug
+
+;
+;[*]
+;
+;  If specified, the '*' provisioning is used for all devices which do not
+;  have another provisioning entry within the file.  If unspecified, no
+;  provisioning will take place for devices which have no entry.  DO NOT
+;  USE A '*' PROVISIONING ENTRY UNLESS YOU KNOW WHAT YOU'RE DOING.
+;
+;template=default

logger.conf

@@ -16,7 +16,7 @@
 ; a fractional second parameter which may be used in this field.  Use %1q
 ; for tenths, %2q for hundredths, etc.
 ;
-dateformat=%F %T       ; ISO 8601 date format
+;dateformat=%F %T       ; ISO 8601 date format
 ;dateformat=%F %T.%3q   ; with milliseconds
 ;
 ;

musiconhold.conf

@@ -47,7 +47,7 @@
 
 [default]
 mode=files
-directory=moh
+directory=/etc/asterisk/media/moh/shop
 
 [shop]
 mode=files

oss.conf

@@ -29,7 +29,7 @@
     autoanswer = yes		; no autoanswer on call
     autohangup = yes		; hangup when other party closes
     ; extension = s		; default extension to call
-    ; context = default		; default context for outgoing calls
+    context = local		; default context for outgoing calls
     ; language = ""		; default language
 
     ; If you set overridecontext to 'yes', then the whole dial string

queues.conf

@@ -179,7 +179,7 @@ monitor-type = MixMonitor
 ; The default value for timeoutpriority is "app" since this was how previous versions of
 ; Asterisk behaved.
 ;
-timeout = 40
+;timeout = 15
 ;retry = 5
 ;timeoutpriority = app|conf
 ;
@@ -561,19 +561,12 @@ timeout = 40
 ;member => Local/1000@default,0,John Smith,SIP/1000
 ;member => Local/2000@default,0,Lorem Ipsum,SIP/2000,no
 
-[gott]
-musicclass=gott
-strategy=ringall
-announce-position=no
-announce-to-first-user=no
-member => IAX2/stubnitz/24
-
 [shop]
 musicclass=shop
 strategy=ringall
 announce-position=no
 announce-to-first-user=no
-member => IAX2/stubnitz/23
+member => SIP/23
 
 [auction]
 musicclass=shop

show-bid.sh

@@ -1,6 +0,0 @@
-#!/bin/sh
-
-echo "Content-Type: text/plain"
-echo ""
-
-asterisk -x 'queue show auction' | sed -n 's/^.*prio: \(.*\))$/Aktuelles Gebot: \1/p' | head -n 1

sip.conf

@@ -818,8 +818,10 @@ qualify=yes
 ; Examples:
 ;
 ;register => 1234:password@mysipprovider.com
-register => 7467:12zX22bH02q2@eventphone.de
+
-register => 7467:e5JvEVLQN9ss@voip.eventphone.de
+;register => 7467:12zX22bH02q2@eventphone.de
+;register => 7467:COwcNVlnErGf@voip.eventphone.de
+
 ;
 ;     This will pass incoming calls to the 's' extension
 ;
@@ -1017,7 +1019,8 @@ registerattempts=0            ; Number of registration attempts before we give u
 ; This does not really work well in the case where Asterisk is outside and the
 ; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
 ;
-;directmedia=yes                ; Asterisk by default tries to redirect the
+directmedia=no
+				; Asterisk by default tries to redirect the
                                 ; RTP media stream to go directly from
                                 ; the caller to the callee.  Some devices do not
                                 ; support this (especially if one of them is behind a NAT).
@@ -1621,30 +1624,32 @@ directrtpsetup=no               ; Enable the new experimental direct RTP setup.
                                 ; external IP address of the remote device. If port forwarding is done at the client side
                                 ; then UDPTL will flow to the remote device.
 
-[eventphone]
+;[eventphone]
-type=friend
+;type=friend
-host=148.251.63.154
+;host=148.251.63.154
-defaultuser=7467
+;defaultuser=7467
-secret=12zX22bH02q2
+;secret=12zX22bH02q2
-context=from-eventphone
+;context=from-eventphone
-qualify=yes
+;qualify=yes
-disallow=all
+;disallow=all
-allow=alaw
+;allow=alaw
-directmedia=no
+;direct-media=no
-dtmfmode=rfc2833
+;canrewrite=no
-insecure=invite,port
+;dtmfmode=rfc2833
-qualify=yes
+;insecure=invite,port
-
+;qualify=yes
-[eventphone-event]
+;
-type=friend
+;[eventphone-event]
-host=93.180.79.52
+;type=friend
-defaultuser=7467
+;host=93.180.79.35
-secret=e5JvEVLQN9ss
+;defaultuser=7467
-context=from-eventphone
+;secret=COwcNVlnErGf
-qualify=yes
+;context=from-eventphone
-disallow=all
+;qualify=yes
-allow=alaw
+;disallow=all
-directmedia=no
+;allow=alaw
-dtmfmode=rfc2833
+;direct-media=no
-insecure=invite,port
+;canrewrite=no
-qualify=yes
+;dtmfmode=rfc2833
+;insecure=invite,port
+;qualify=yes

static-http/cgi-bin/bid

@@ -1,3 +0,0 @@
-#!/bin/sh
-
-exec sudo /etc/asterisk/show-bid.sh

static-http/queueview.js

@@ -1,5 +1,5 @@
 const api_creds = "queueview:queueview";
-const api_app = "queueview_" + (""+Math.random()).substring(2);
+const api_app = "queueview"; //_" + (""+Math.random()).substring(2);
 
 let ws;
 

users.conf

@@ -126,3 +126,8 @@ context=local
 secret=teleshop23
 host=dynamic
 context=local
+
+[24]
+secret=teleshop24
+host=dynamic
+context=local