Ignore swap files

We don't want to service this to the Interwebz

.gitignore

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

asound.state

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

clear-queue-members.sh

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

extensions.conf

@@ -732,15 +732,20 @@ 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,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)
@@ -762,14 +767,15 @@ 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),Set(__AUCTION_BID=${money})
+same  => n(direct_queue),GotoIf($[ "${DB(auction/open)}" != "1" ]?bidding-closed,1)
+same  => n,Set(__AUCTION_BID=${money})
 same  => n,Queue(auction,tc)
-same  => n,Dial(SIP/21,60,U(bidding-announce))
+same  => n,Dial(IAX2/stubnitz/21,60,U(bidding-announce))
 same  => n,Hangup()
 
 exten => bidding-closed,1,Playback(/etc/asterisk/media/playback/e_5_leider_kann_gerade_kein_gebot)
@@ -781,32 +787,41 @@ same  => n,Goto(1,1)
 
 [auction-spy]
 exten => s,1,Noop()
-same  => n,ChanSpy(SIP/22,bqv(2))
+same  => n,ChanSpy(SIP/22,bq)
 
 [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(SIP/${EXTEN},,U(bidding-announce))
+same  => n,Dial(IAX2/stubnitz/${EXTEN},,U(bidding-announce))
+same  => n,RemoveQueueMember(auction,Local/${EXTEN}@bidding-queue-member)
+same  => n,Hangup
 
 [take-question]
-exten => s,1,Playback(/etc/asterisk/media/playback/e_4_zur_qualit_tssicherung)
+exten => s,1,Background(/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
 
-exten => _X.,1,Dial(IAX2/hetzner/${EXTEN})
-same  => n,Hangup
-
-[oldlocal]
+[fromstubnitz]
+exten => _22X,1,Dial(IAX2/stubnitz/${EXTEN:1})
 exten => _0X.,1,Goto(to-eventphone,${EXTEN:1},1)
 
 ; Open bidding, read start bid
@@ -847,11 +862,13 @@ 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=0.0.0.0
+bindaddr=172.19.27.2
 ;
 ; 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=172.19.27.1    ; You can specify 'bindaddr' more than once to bind to
+bindaddr=172.19.27.2    ; You can specify 'bindaddr' more than once to bind to
                          ; multiple addresses, but the first will be the
                          ; default. IPv6 addresses are accepted.
 
@@ -149,11 +149,12 @@ bandwidth=low
 ; You can also fine tune codecs here using "allow" and "disallow" clauses with
 ; specific codecs.  Use "all" to represent all formats.
 ;
-disallow=all
-allow=gsm
+;allow=all
 ;disallow=g723.1
 ;disallow=lpc10
 ;allow=gsm
+disallow=all
+allow=gsm
 ;
 
 ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
@@ -481,9 +482,9 @@ autokill=yes
 ;
 ;shrinkcallerid=yes     ; on by default
 
-;; Guest sections for unauthenticated connection attempts.  Just specify an
-;; empty secret, or provide no secret section.
-;;
+; Guest sections for unauthenticated connection attempts.  Just specify an
+; empty secret, or provide no secret section.
+;
 ;[guest]
 ;type=user
 ;context=public
@@ -669,12 +670,12 @@ autokill=yes
 ;context=number-please ; we start at the s extension in this context
 ;
 
-[hetzner]
+[stubnitz]
 type=friend
-host=172.19.27.2
-username=stubnitz
+host=172.19.27.1
+username=hetzner
 secret=f9DuF9Sh0GRKUFrF
-context=fromhetzner
+context=fromstubnitz
 qualify=yes
 qualifyfreqok=10000
 qualifyfreqnotok=10000

iaxprov.conf

@@ -1,80 +0,0 @@
-;
-; 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

musiconhold.conf

@@ -47,7 +47,7 @@
 
 [default]
 mode=files
-directory=/etc/asterisk/media/moh/shop
+directory=moh
 
 [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 = local		; default context for outgoing calls
+    ; context = default		; 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 = 15
+timeout = 40
 ;retry = 5
 ;timeoutpriority = app|conf
 ;
@@ -561,12 +561,19 @@ monitor-type = MixMonitor
 ;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 => SIP/23
+member => IAX2/stubnitz/23
 
 [auction]
 musicclass=shop

show-bid.sh

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

sip.conf

@@ -818,10 +818,8 @@ qualify=yes
 ; Examples:
 ;
 ;register => 1234:password@mysipprovider.com
-
-;register => 7467:12zX22bH02q2@eventphone.de
-;register => 7467:COwcNVlnErGf@voip.eventphone.de
-
+register => 7467:12zX22bH02q2@eventphone.de
+register => 7467:COwcNVlnErGf@voip.eventphone.de
 ;
 ;     This will pass incoming calls to the 's' extension
 ;
@@ -1019,8 +1017,7 @@ 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=no
-				; Asterisk by default tries to redirect the
+;directmedia=yes                ; 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).
@@ -1057,7 +1054,7 @@ directmedia=no
                                 ; the option in this situation helps to prevent potential glares.
                                 ; Setting this option implies 'yes'.
 
-directrtpsetup=no             ; Enable the new experimental direct RTP setup. This sets up
+directrtpsetup=no               ; Enable the new experimental direct RTP setup. This sets up
                                 ; the call directly with media peer-2-peer without re-invites.
                                 ; Will not work for video and cases where the callee sends
                                 ; RTP payloads and fmtp headers in the 200 OK that does not match the
@@ -1624,32 +1621,30 @@ directrtpsetup=no             ; Enable the new experimental direct RTP setup. Th
                                 ; 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]
-;type=friend
-;host=148.251.63.154
-;defaultuser=7467
-;secret=12zX22bH02q2
-;context=from-eventphone
-;qualify=yes
-;disallow=all
-;allow=alaw
-;direct-media=no
-;canrewrite=no
-;dtmfmode=rfc2833
-;insecure=invite,port
-;qualify=yes
-;
-;[eventphone-event]
-;type=friend
-;host=93.180.79.35
-;defaultuser=7467
-;secret=COwcNVlnErGf
-;context=from-eventphone
-;qualify=yes
-;disallow=all
-;allow=alaw
-;direct-media=no
-;canrewrite=no
-;dtmfmode=rfc2833
-;insecure=invite,port
-;qualify=yes
+[eventphone]
+type=friend
+host=148.251.63.154
+defaultuser=7467
+secret=12zX22bH02q2
+context=from-eventphone
+qualify=yes
+disallow=all
+allow=alaw
+directmedia=no
+dtmfmode=rfc2833
+insecure=invite,port
+qualify=yes
+
+[eventphone-event]
+type=friend
+host=93.180.79.35
+defaultuser=7467
+secret=COwcNVlnErGf
+context=from-eventphone
+qualify=yes
+disallow=all
+allow=alaw
+directmedia=no
+dtmfmode=rfc2833
+insecure=invite,port
+qualify=yes

static-http/cgi-bin/bid

@@ -0,0 +1,3 @@
+#!/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

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