Add SIP station 24 for telegott line

This reverts commit a6b9caad5a.

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 {

extensions.conf

@@ -781,7 +781,7 @@ 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)
@@ -801,7 +801,12 @@ same  => n,Hangup()
 
 [local]
 exten => _22X,1,Dial(SIP/${EXTEN:1})
+same  => n,Hangup
 
+exten => _X.,1,Dial(IAX2/hetzner/${EXTEN})
+same  => n,Hangup
+
+[oldlocal]
 exten => _0X.,1,Goto(to-eventphone,${EXTEN:1},1)
 
 ; Open bidding, read start bid
@@ -845,3 +850,8 @@ 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=0.0.0.0
 ;
 ; Port to bind to for HTTP sessions (default is 8088)
 ;

iax.conf

@@ -0,0 +1,680 @@
+;
+; Inter-Asterisk eXchange v2 (IAX2) Channel Driver configuration
+;
+; This configuration is read when the chan_iax2.so module is loaded, and is
+; re-read when the module is reloaded, such as when invoking the CLI command:
+;
+;     *CLI> iax2 reload
+;
+
+; General settings, like port number to bind to, and an option address (the
+; default is to bind to all local addresses).
+
+[general]
+
+; Listener Addresses
+;
+; Use the 'bindaddr' and 'bindport' options to specify on which address and port
+; the IAX2 channel driver will listen for incoming requests.
+;
+;
+
+;bindport=4569           ; The default port to listen on
+                         ; NOTE: bindport must be specified BEFORE bindaddr or
+                         ; may be specified on a specific bindaddr if followed by
+                         ; colon and port (e.g. bindaddr=192.168.0.1:4569) or for
+                         ; 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
+                         ; multiple addresses, but the first will be the
+                         ; default. IPv6 addresses are accepted.
+
+;
+; Set 'iaxcompat' to yes if you plan to use layered switches or some other
+; scenario which may cause some delay when doing a lookup in the dialplan. It
+; incurs a small performance hit to enable it. This option causes Asterisk to
+; spawn a separate thread when it receives an IAX2 DPREQ (Dialplan Request)
+; instead of blocking while it waits for a response.
+;
+; Accepted values: yes, no
+; Default value:   no
+;
+;iaxcompat=yes
+;
+
+;
+; Disable UDP checksums (if nochecksums is set, then no checkums will
+; be calculated/checked on systems supporting this feature)
+;
+; Accepted values: yes, no
+; Default value:   no
+;
+;nochecksums=yes
+;
+
+;
+; For increased security against brute force password attacks enable
+; 'delayreject' which will delay the sending of authentication reject for REGREQ
+; or AUTHREP if there is a password.
+;
+; Accepted values: yes, no
+; Default value:   no
+;
+;delayreject=yes
+;
+
+;
+; You may specify a global default AMA flag for iaxtel calls.  These flags are
+; used in the generation of call detail records.
+;
+; Accepted values: default, omit, billing, documentation
+; Default value:   default
+;
+;amaflags=billing
+;
+
+;
+; ADSI (Analog Display Services Interface) can be enabled if you have (or may
+; have) ADSI compatible CPE equipment.
+;
+; Accepted values: yes, no
+; Default value:   no
+;
+;adsi=yes
+;
+
+;
+; Whether or not to perform an SRV lookup on outbound calls.
+;
+; Accepted values: yes, no
+; Default value:   no
+;
+;srvlookup=yes
+;
+
+;
+; You may specify a default account for Call Detail Records (CDRs) in addition to
+; specifying on a per-user basis.
+;
+; Accepted values: Any string value up to 19 characters in length
+; Default value:   <empty>
+;
+;accountcode=lss0101
+;
+
+;
+; You may specify a global default language for users.  This can be specified
+; also on a per-user basis.  If omitted, will fallback to English (en).
+;
+; Accepted values: A language tag such as 'en' or 'es'
+; Default value:   en
+;
+;language=en
+;
+
+;
+; This option specifies a preference for which music-on-hold class this channel
+; should listen to when put on hold if the music class has not been set on the
+; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
+; channel putting this one on hold did not suggest a music class.
+;
+; If this option is set to "passthrough", then the hold message will always be
+; passed through as signalling instead of generating hold music locally.
+;
+; This option may be specified globally, or on a per-user or per-peer basis.
+;
+; Accepted values: passthrough, or any music-on-hold class name
+; Default value:   <empty>
+;
+;mohinterpret=default
+;
+
+;
+; The 'mohsuggest' 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.
+;
+;mohsuggest=default
+;
+
+;
+; Specify bandwidth of low, medium, or high to control which codecs are used
+; in general.
+;
+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
+;disallow=g723.1
+;disallow=lpc10
+;allow=gsm
+;
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+; Jitter Buffer
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+;
+; You can adjust several parameters relating to the jitter buffer.  The jitter
+; buffer's function is to compensate for varying network delay.
+;
+; All of the jitter buffer settings are in milliseconds.  The jitter buffer
+; works for INCOMING audio only - the outbound audio will be dejittered by the
+; jitter buffer at the other end.
+;
+; jitterbuffer=yes|no: global default as to whether you want
+; the jitter buffer at all.
+;
+; maxjitterbuffer: a maximum size for the jitter buffer.
+; Setting a reasonable maximum here will prevent the call delay
+; from rising to silly values in extreme situations; you'll hear
+; SOMETHING, even though it will be jittery.
+;
+; resyncthreshold: when the jitterbuffer notices a significant change in delay
+; that continues over a few frames, it will resync, assuming that the change in
+; delay was caused by a timestamping mix-up. The threshold for noticing a
+; change in delay is measured as twice the measured jitter plus this resync
+; threshold.
+; Resyncing can be disabled by setting this parameter to -1.
+;
+; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer
+; should return in a row. Since some clients do not send CNG/DTX frames to
+; indicate silence, the jitterbuffer will assume silence has begun after
+; returning this many interpolations. This prevents interpolating throughout
+; a long silence.
+;
+; jittertargetextra: 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.
+;
+
+jitterbuffer=no
+;maxjitterbuffer=1000
+;maxjitterinterps=10
+;resyncthreshold=1000
+;jittertargetextra=40
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+; IAX2 Encryption
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+;
+; Enable IAX2 encryption.  The default is no.
+;
+;encryption=yes
+;
+
+;
+; Force encryption insures no connection is established unless both sides
+; support encryption.  By turning this option on, encryption is automatically
+; turned on as well.  The default is no.
+;
+;forceencryption=yes
+;
+
+; This option defines the maximum payload in bytes an IAX2 trunk can support at
+; a given time.  The best way to explain this is to provide an example.  If the
+; maximum number of calls to be supported is 800, and each call transmits 20ms
+; frames of audio using ulaw:
+;
+;     (8000hz / 1000ms) * 20ms * 1 byte per sample = 160 bytes per frame
+;
+; The maximum load in bytes is:
+;
+;     (160 bytes per frame) * (800 calls) = 128000 bytes
+;
+; Once this limit is reached, calls may be dropped or begin to lose audio.
+; Depending on the codec in use and number of channels to be supported this value
+; may need to be raised, but in most cases the default value is large enough.
+;
+; trunkmaxsize = 128000 ; defaults to 128000 bytes, which supports up to 800
+                        ; calls of ulaw at 20ms a frame.
+
+; With a large amount of traffic on IAX2 trunks, there is a risk of bad voice
+; quality when allowing the Linux system to handle fragmentation of UDP packets.
+; Depending on the size of each payload, allowing the OS to handle fragmentation
+; may not be very efficient. This setting sets the maximum transmission unit for
+; IAX2 UDP trunking. The default is 1240 bytes which means if a trunk's payload
+; is over 1240 bytes for every 20ms it will be broken into multiple 1240 byte
+; messages.  Zero disables this functionality and let's the OS handle
+; fragmentation.
+;
+; trunkmtu = 1240    ; trunk data will be sent in 1240 byte messages.
+
+; trunkfreq sets how frequently trunk messages are sent in milliseconds. This
+; value is 20ms by default, which means the trunk will send all the data queued
+; to it in the past 20ms.  By increasing the time between sending trunk messages,
+; the trunk's payload size will increase as well.  Note, depending on the size
+; set by trunkmtu, messages may be sent more often than specified.  For example
+; if a trunk's message size grows to the trunkmtu size before 20ms is reached
+; that message will be sent immediately.  Acceptable values are between 10ms and
+; 1000ms.
+;
+; trunkfreq=20    ; How frequently to send trunk msgs (in ms). This is 20ms by
+                  ; default.
+
+; Should we send timestamps for the individual sub-frames within trunk frames?
+; There is a small bandwidth use for these (less than 1kbps/call), but they
+; ensure that frame timestamps get sent end-to-end properly.  If both ends of
+; all your trunks go directly to TDM, _and_ your trunkfreq equals the frame
+; length for your codecs, you can probably suppress these.  The receiver must
+; also support this feature, although they do not also need to have it enabled.
+;
+; trunktimestamps=yes
+
+; Minimum and maximum amounts of time that IAX2 peers can request as a
+; registration expiration interval (in seconds).
+; minregexpire = 60
+; maxregexpire = 60
+
+; IAX2 helper threads
+
+; Establishes the number of iax helper threads to handle I/O.
+; iaxthreadcount = 10
+
+; Establishes the number of extra dynamic threads that may be spawned to handle I/O
+; iaxmaxthreadcount = 100
+
+;
+; We can register with another IAX2 server to let him know where we are
+; in case we have a dynamic IP address for example
+;
+; Register with tormenta using username marko and password secretpass
+;
+;register => marko:secretpass@tormenta.linux-support.net
+;
+; Register joe at remote host with no password
+;
+;register => joe@remotehost:5656
+;
+; Register marko at tormenta.linux-support.net using RSA key "torkey"
+;
+;register => marko:[torkey]@tormenta.linux-support.net
+;
+; Sample Registration for iaxtel
+;
+; Visit http://www.iaxtel.com to register with iaxtel.  Replace "user"
+; and "pass" with your username and password for iaxtel.  Incoming
+; calls arrive at the "s" extension of "default" context.
+;
+;register => user:pass@iaxtel.com
+;
+; Sample Registration for IAX2 + FWD
+;
+; To register using IAX2 with FWD, it must be enabled by visiting the URL
+; http://www.fwdnet.net/index.php?section_id=112
+;
+; Note that you need an extension in you default context which matches
+; your free world dialup number.  Please replace "FWDNumber" with your
+; FWD number and "passwd" with your password.
+;
+;register => FWDNumber:passwd@iax.fwdnet.net
+;
+; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
+; perceived external network address has changed.  When the stun_monitor is installed and
+; configured, chan_iax will renew all outbound registrations when the monitor detects any sort
+; of network change has occurred. By default this option is enabled, but only takes effect once
+; res_stun_monitor is configured.  If res_stun_monitor is enabled and you wish to not
+; generate all outbound registrations on a network change, use the option below to disable
+; this feature.
+;
+; subscribe_network_change_event = yes ; on by default
+;
+; You can enable authentication debugging to increase the amount of
+; debugging traffic.
+;
+;authdebug = yes
+;
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
+;tos=ef
+;cos=5
+;
+; If regcontext is specified, Asterisk will dynamically create and destroy
+; a NoOp priority 1 extension for a given peer who registers or unregisters
+; with us.  The actual extension is the 'regexten' parameter of the registering
+; peer or its name if 'regexten' is not provided.  More than one regexten
+; may be supplied if they are separated by '&'.  Patterns may be used in
+; regexten.
+;
+;regcontext=iaxregistrations
+;
+; If we don't get ACK to our NEW within 2000ms, and autokill is set to yes,
+; then we cancel the whole thing (that's enough time for one retransmission
+; only).  This is used to keep things from stalling for a long time for a host
+; that is not available, but would be ill advised for bad connections.  In
+; addition to 'yes' or 'no' you can also specify a number of milliseconds.
+; See 'qualify' for individual peers to turn on for just a specific peer.
+;
+autokill=yes
+;
+; codecpriority controls the codec negotiation of an inbound IAX2 call.
+; This option is inherited to all user entities.  It can also be defined
+; in each user entity separately which will override the setting in general.
+;
+; The valid values are:
+;
+; caller   - Consider the callers preferred order ahead of the host's.
+; host     - Consider the host's preferred order ahead of the caller's.
+; disabled - Disable the consideration of codec preference altogether.
+;            (this is the original behaviour before preferences were added)
+; reqonly  - Same as disabled, only do not consider capabilities if
+;            the requested format is not available the call will only
+;            be accepted if the requested format is available.
+;
+; The default value is 'host'
+;
+;codecpriority=host
+;
+; allowfwdownload controls whether this host will serve out firmware to
+; IAX2 clients which request it.  This has only been used for the IAXy,
+; and it has been recently proven that this firmware distribution method
+; can be used as a source of traffic amplification attacks.  Also, the
+; IAXy firmware has not been updated for at least 18 months, so unless
+; you are provisioning IAXys in a secure network, we recommend that you
+; leave this option to the default, off.
+;
+;allowfwdownload=yes
+
+;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
+                    ; just like friends added from the config file only on a
+                    ; as-needed basis? (yes|no)
+
+;rtsavesysname=yes  ; Save systemname in realtime database at registration
+                    ; Default = no
+
+;rtupdate=yes       ; Send registry updates to database using realtime? (yes|no)
+                    ; If set to yes, when a IAX2 peer registers successfully,
+                    ; the IP address, the origination port, the registration period,
+                    ; and the username of the peer will be set to database via realtime.
+                    ; If not present, defaults to 'yes'.
+
+;rtautoclear=yes    ; Auto-Expire friends created on the fly on the same schedule
+                    ; as if it had just registered? (yes|no|<seconds>)
+                    ; If set to yes, when the registration expires, the friend will
+                    ; vanish from the configuration until requested again.
+                    ; If set to an integer, friends expire within this number of
+                    ; seconds instead of the registration interval.
+
+;rtignoreregexpire=yes ; When reading a peer from Realtime, if the peer's registration
+                       ; has expired based on its registration interval, used the stored
+                       ; address information regardless. (yes|no)
+
+;parkinglot=edvina     ; Default parkinglot for IAX2 peers and users
+                       ; This can also be configured per device
+                       ; Parkinglots are defined in features.conf
+
+;
+; The following two options are used to disable call token validation for the
+; purposes of interoperability with IAX2 endpoints that do not yet support it.
+;
+; Call token validation can be set as optional for a single IP address or IP
+; address range by using the 'calltokenoptional' option. 'calltokenoptional' is
+; only a global option.
+;
+;calltokenoptional=209.16.236.73/255.255.255.0
+;
+; By setting 'requirecalltoken=no', call token validation becomes optional for
+; that peer/user.  By setting 'requirecalltoken=auto', call token validation
+; is optional until a call token supporting peer registers successfully using
+; call token validation.  This is used as an indication that from now on, we
+; can require it from this peer.  So, requirecalltoken is internally set to yes.
+; requirecalltoken may only be used in peer/user/friend definitions,
+; not in the global scope.
+; By default, 'requirecalltoken=yes'.
+;
+;requirecalltoken=no
+;
+; Maximum time allowed for call token authentication handshaking. Default is 10 seconds.
+; Use higher values in lagged or high packet loss networks.
+;
+;calltokenexpiration=10
+
+;
+; These options are used to limit the amount of call numbers allocated to a
+; single IP address.  Before changing any of these values, it is highly encouraged
+; to read the user guide associated with these options first.  In most cases, the
+; default values for these options are sufficient.
+;
+; The 'maxcallnumbers' option limits the amount of call numbers allowed for each
+; individual remote IP address.  Once an IP address reaches it's call number
+; limit, no more new connections are allowed until the previous ones close.  This
+; option can be used in a peer definition as well, but only takes effect for
+; the IP of a dynamic peer after it completes registration.
+;
+;maxcallnumbers=512
+;
+; The 'maxcallnumbers_nonvalidated' is used to set the combined number of call
+; numbers that can be allocated for connections where call token  validation
+; has been disabled.  Unlike the 'maxcallnumbers' option, this limit is not
+; separate for each individual IP address.  Any connection resulting in a
+; non-call token validated call number being allocated contributes to this
+; limit.  For use cases, see the call token user guide.  This option's
+; default value of 8192 should be sufficient in most cases.
+;
+;maxcallnumbers_nonvalidated=1024
+;
+; The [callnumberlimits] section allows custom call number limits to be set
+; for specific IP addresses and IP address ranges.  These limits take precedence
+; over the global 'maxcallnumbers' option, but may still be overridden by a
+; peer defined 'maxcallnumbers' entry.  Note that these limits take effect
+; for every individual address within the range, not the range as a whole.
+;
+;[callnumberlimits]
+;10.1.1.0/255.255.255.0 = 24
+;10.1.2.0/255.255.255.0 = 32
+;
+
+; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
+; in square brackets.  For example, the Caller*ID value 555.5555 becomes 5555555
+; when this option is enabled.  Disabling this option results in no modification
+; of the Caller*ID value, which is necessary when the Caller*ID represents something
+; that must be preserved.  This option can only be used in the [general] section.
+; By default this option is on.
+;
+;shrinkcallerid=yes     ; on by default
+
+;; Guest sections for unauthenticated connection attempts.  Just specify an
+;; empty secret, or provide no secret section.
+;;
+;[guest]
+;type=user
+;context=public
+;callerid="Guest IAX User"
+;
+;;
+;; Trust Caller*ID Coming from iaxtel.com
+;;
+;[iaxtel]
+;type=user
+;context=default
+;auth=rsa
+;inkeys=iaxtel
+;
+;;
+;; Trust Caller*ID Coming from iax.fwdnet.net
+;;
+;[iaxfwd]
+;type=user
+;context=default
+;auth=rsa
+;inkeys=freeworlddialup
+;
+;
+; Trust Caller*ID delivered over DUNDi/e164
+;
+;[dundi]
+;type=user
+;dbsecret=dundi/secret
+;context=dundi-e164-local
+
+;
+; Further user sections may be added, specifying a context and a secret used
+; for connections with that given authentication name.  Limited IP based
+; access control is allowed by use of "permit", "deny", and "acl" keywords.
+; Multiple rules are permitted. Multiple permitted contexts may be specified,
+; in which case the first will be the default.  You can also override
+; Caller*ID so that when you receive a call you set the Caller*ID to be what
+; you want instead of trusting what the remote user provides
+;
+; There are three authentication methods that are supported:  md5, plaintext,
+; and rsa.  The least secure is "plaintext", which sends passwords cleartext
+; across the net.  "md5" uses a challenge/response md5 sum arrangement, but
+; still requires both ends have plain text access to the secret.  "rsa" allows
+; unidirectional secret knowledge through public/private keys.  If "rsa"
+; authentication is used, "inkeys" is a list of acceptable public keys on the
+; local system that can be used to authenticate the remote peer, separated by
+; the ":" character.  "outkey" is a single, private key to use to authenticate
+; to the other side.  Public keys are named /var/lib/asterisk/keys/<name>.pub
+; while private keys are named /var/lib/asterisk/keys/<name>.key.  Private
+; keys should always be 3DES encrypted.
+;
+;
+; NOTE: All hostnames and IP addresses in this file are for example purposes
+;       only; you should not expect any of them to actually be available for
+;       your use.
+;
+;[markster]
+;type=user
+;context=default
+;context=local
+;auth=md5,plaintext,rsa
+;secret=markpasswd
+;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.
+;dbsecret=mysecrets/place    ; Secrets can be stored in astdb, too
+;transfer=no                 ; Disable IAX2 native transfer
+;transfer=mediaonly          ; When doing IAX2 native transfers, transfer only
+                             ; the media stream
+;jitterbuffer=yes            ; Override the global setting and enable the jitter
+                             ; buffer for this user
+;maxauthreq=10               ; Set the maximum number of outstanding AUTHREQs
+                             ; waiting for replies. If this limit is reached,
+                             ; any further authentication will be blocked, until
+                             ; the pending requests expire or a reply is
+                             ; received.
+;callerid="Mark Spencer" <(256) 428-6275>
+;deny=0.0.0.0/0.0.0.0
+;accountcode=markster0101
+;permit=209.16.236.73/255.255.255.0
+;language=en                 ; Use english as default language
+;encryption=yes              ; Enable IAX2 encryption.  The default is no.
+;keyrotate=off               ; This is a compatibility option for older versions
+                             ; of IAX2 that do not support key rotation with
+                             ; encryption.  This option will disable the
+                             ; IAX_COMMAND_RTENC message.  The default is on.
+
+;
+; Peers may also be specified, with a secret and a remote hostname.
+;
+;[demo]
+;type=peer
+;username=asterisk
+;secret=supersecret
+;host=192.168.10.10
+;description=My IAX2 Peer            ; Description of this peer, as listed by
+                                     ; 'iax2 show peers'
+;sendani=no
+;host=asterisk.linux-support.net
+;port=5036
+;mask=255.255.255.255
+;qualify=yes                ; Make sure this peer is alive.
+;qualifysmoothing = yes     ; Use an average of the last two PONG results to
+                            ; reduce falsely detected LAGGED hosts.  The default
+                            ; is 'no.'
+;qualifyfreqok = 60000      ; How frequently to ping the peer when everything
+                            ; seems to be OK, in milliseconds.
+;qualifyfreqnotok = 10000   ; How frequently to ping the peer when it's either
+                            ; LAGGED or UNAVAILABLE, in milliseconds.
+;jitterbuffer=no            ; Turn off jitter buffer for this peer
+;
+;encryption=yes             ; Enable IAX2 encryption.  The default is no.
+;keyrotate=off              ; This is a compatibility option for older versions
+                            ; of IAX2 that do not support key rotation with
+                            ; encryption.  This option will disable the
+                            ; IAX_COMMAND_RTENC message.  The default is 'on.'
+
+; Peers can remotely register as well, so that they can be mobile.  Default
+; IPs can also optionally be given but are not required.  Caller*ID can be
+; suggested to the other side as well if it is for example a phone instead of
+; another PBX.
+;connectedline=yes ; Set if connected line and redirecting information updates
+;                  ; are passed between Asterisk servers for this peer.
+;                  ; yes - Sending and receiving updates are enabled.
+;                  ; send - Only send updates.
+;                  ; receive - Only process received updates.
+;                  ; no - Sending and receiving updates are disabled.
+;                  ; Default is "no".
+;                  ;
+;                  ; Note: Because of an incompatibility between Asterisk v1.4
+;                  ; and Asterisk v1.8 or later, this option must be set
+;                  ; to "no" toward the Asterisk v1.4 peer.  A symptom of the
+;                  ; incompatibility is the call gets disconnected unexpectedly.
+
+
+;[dynamichost]
+;host=dynamic
+;secret=mysecret
+; Note: app_voicemail mailboxes must be in the form of mailbox@context.
+;mailbox=1234		; Notify about mailbox 1234
+;inkeys=key1:key2
+;peercontext=local	; Default context to request for calls to peer
+;defaultip=216.207.245.34
+;callerid="Some Host" <(256) 428-6011>
+
+;[biggateway]
+;type=peer
+;host=192.168.0.1
+;description=Gateway to PSTN
+;context=*
+;secret=myscret
+;trunk=yes			; Use IAX2 trunking with this host
+;timezone=America/New_York	; Set a timezone for the date/time IE
+
+;
+; Friends are a shortcut for creating a user and a peer with the same values.
+;
+
+;[marko]
+;type=friend
+;host=dynamic
+;regexten=1234
+;secret=moofoo   ; Multiple secrets may be specified. For a "user", all
+;secret=foomoo   ; specified entries will be accepted as valid. For a "peer",
+;secret=shazbot  ; only the last specified secret will be used.
+;context=default
+;permit=0.0.0.0/0.0.0.0
+;acl=example_named_acl
+
+;
+; With immediate=yes, an IAX2 phone or a phone on an IAXy acts as a hot-line
+; which goes immediately to the s extension when picked up.  Useful for
+; elevator phones, manual service, or other similar applications.
+;
+;[manual]
+;type=friend
+;host=dynamic
+;immediate=yes  ; go immediately to s extension when picked up
+;secret=moofoo	; when immediate=yes is specified, secret is required
+;context=number-please ; we start at the s extension in this context
+;
+
+[hetzner]
+type=friend
+host=172.19.27.2
+username=stubnitz
+secret=f9DuF9Sh0GRKUFrF
+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

musiconhold.conf

@@ -47,11 +47,15 @@
 
 [default]
 mode=files
-directory=moh
+directory=/etc/asterisk/media/moh/shop
 
 [shop]
 mode=files
 directory=/etc/asterisk/media/moh/shop
+
+[gott]
+mode=files
+directory=/etc/asterisk/media/moh/gott
 ;
 ;[native-random]
 ;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

sip.conf

@@ -818,8 +818,10 @@ 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
 ;
@@ -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).
@@ -1054,7 +1057,7 @@ registerattempts=0            ; Number of registration attempts before we give u
                                 ; the option in this situation helps to prevent potential glares.
                                 ; Setting this option implies 'yes'.
 
-;directrtpsetup=yes             ; 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
@@ -1621,32 +1624,32 @@ registerattempts=0            ; Number of registration attempts before we give u
                                 ; 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
-direct-media=no
+;direct-media=no
-canrewrite=no
+;canrewrite=no
-dtmfmode=rfc2833
+;dtmfmode=rfc2833
-insecure=invite,port
+;insecure=invite,port
-qualify=yes
+;qualify=yes
-
+;
-[eventphone-event]
+;[eventphone-event]
-type=friend
+;type=friend
-host=93.180.79.35
+;host=93.180.79.35
-defaultuser=7467
+;defaultuser=7467
-secret=COwcNVlnErGf
+;secret=COwcNVlnErGf
-context=from-eventphone
+;context=from-eventphone
-qualify=yes
+;qualify=yes
-disallow=all
+;disallow=all
-allow=alaw
+;allow=alaw
-direct-media=no
+;direct-media=no
-canrewrite=no
+;canrewrite=no
-dtmfmode=rfc2833
+;dtmfmode=rfc2833
-insecure=invite,port
+;insecure=invite,port
-qualify=yes
+;qualify=yes

users.conf

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