[gelöst] Ringgroup -> Pickup

schwankit

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Hallo Zusammen,

ich habe 2 Extension (A,B)(Pickupgroup 1) in 1 Ringgroup.
1 weitere Extension (C) (Pickupgroup 1) in einer anderen Ringgroup.

Jetzt möchte ich mit Extension (C) wenn Extension A,B (da in der gleichen Ringgroup) klingeln, den anruf holen, bevor A,B abheben.
Wie stelle ich das am geschicktesten an bzw. was muss ich in welche Datei schreiben?

Bin für jede Hilfe dankbar!
 
Zuletzt bearbeitet:
@rentier-s: vielen Dank schonmal aber wie tippe ich das an meinem Telefon ein? *(Ringgroupname) *(Kurzwahl eines Users der Ringgroup) oder wie?

- - - Aktualisiert - - -

Ich drehe mich hier seit Tagen im Kreis :( evtl. findet ja jemand den Fehler in meinen Configs, der sich mir nicht zeigen mag :(

SIP.conf
Code:
;!
;! Automatically generated configuration file
;! Filename: sip.conf (/var/packages/Asterisk/target/etc/asterisk/sip.conf)
;! Generator: Manager
;! Creation Date: Mon Sep 12 09:33:49 2016
;!
;
; SIP Configuration example for Asterisk
;
; Note: Please read the security documentation for Asterisk in order to
;     understand the risks of installing Asterisk with the sample
;    configuration. If your Asterisk is installed on a public
;    IP address connected to the Internet, you will want to learn
;    about the various security settings BEFORE you start
;    Asterisk.
;
;    Especially note the following settings:
;        - allowguest (default enabled)
;        - permit/deny/acl - IP address filters
;        - contactpermit/contactdeny/contactacl - IP address filters for registrations
;        - context - Which set of services you offer various users
;
; SIP dial strings
;-----------------------------------------------------------
; In the dialplan (extensions.conf) you can use several
; syntaxes for dialing SIP devices.
;        SIP/devicename
;        SIP/username@domain   (SIP uri)
;        SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
;        SIP/devicename/extension
;        SIP/devicename/extension/IPorHost
;        SIP/username@domain//IPorHost
;
;
; Devicename
;        devicename is defined as a peer in a section below.
;
; username@domain
;        Call any SIP user on the Internet
;        (Don't forget to enable DNS SRV records if you want to use this)
;
; devicename/extension
;        If you define a SIP proxy as a peer below, you may call
;        SIP/proxyhostname/user or SIP/user@proxyhostname
;        where the proxyhostname is defined in a section below
;        This syntax also works with ATA's with FXO ports
;
; SIP/username[:password[:md5secret[:authname]]]@host[:port]
;        This form allows you to specify password or md5secret and authname
;        without altering any authentication data in config.
;        Examples:
;
;        SIP/*98@mysipproxy
;        SIP/sales:topsecret::[email protected]:5062
;        SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:[email protected]
;
; IPorHost
;        The next server for this call regardless of domain/peer
;
; All of these dial strings specify the SIP request URI.
; In addition, you can specify a specific To: header by adding an
; exclamation mark after the dial string, like
;
;         SIP/sales@[email protected]
;
; A new feature for 1.8 allows one to specify a host or IP address to use
; when routing the call. This is typically used in tandem with func_srv if
; multiple methods of reaching the same domain exist. The host or IP address
; is specified after the third slash in the dialstring. Examples:
;
; SIP/devicename/extension/IPorHost
; SIP/username@domain//IPorHost
;
; CLI Commands
; -------------------------------------------------------------
; Useful CLI commands to check peers/users:
;   sip show peers               Show all SIP peers (including friends)
;   sip show registry            Show status of hosts we register with
;
;   sip set debug on             Show all SIP messages
;
;   sip reload                   Reload configuration file
;   sip show settings            Show the current channel configuration
;
;------- Naming devices ------------------------------------------------------
;
; When naming devices, make sure you understand how Asterisk matches calls
; that come in.
;    1. Asterisk checks the SIP From: address username and matches against
;       names of devices with type=user
;       The name is the text between square brackets [name]
;    2. Asterisk checks the From: addres and matches the list of devices
;       with a type=peer
;    3. Asterisk checks the IP address (and port number) that the INVITE
;       was sent from and matches against any devices with type=peer
;
; Don't mix extensions with the names of the devices. Devices need a unique
; name. The device name is *not* used as phone numbers. Phone numbers are
; anything you declare as an extension in the dialplan (extensions.conf).
;
; When setting up trunks, make sure there's no risk that any From: username
; (caller ID) will match any of your device names, because then Asterisk
; might match the wrong device.
;
; Note: The parameter "username" is not the username and in most cases is
;       not needed at all. Check below. In later releases, it's renamed
;       to "defaultuser" which is a better name, since it is used in
;       combination with the "defaultip" setting.
;-----------------------------------------------------------------------------

; ** Old configuration options **
; The "call-limit" configuation option is considered old is replaced
; by new functionality. To enable callcounters, you use the new 
; "callcounter" setting (for extension states in queue and subscriptions)
; You are encouraged to use the dialplan groupcount functionality
; to enforce call limits instead of using this channel-specific method.
; You can still set limits per device in sip.conf or in a database by using
; "setvar" to set variables that can be used in the dialplan for various limits.

[general]

srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
alifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
alifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
pickupexten = *8
pickupsound = beep
pickupfailsound = beeperr
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
alifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
alifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
alifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
alifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
alifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
alifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
alifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
alifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
pickupexten = *8
pickupsound = beep
pickupfailsound = beeperr
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
alifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
alifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
alifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
alifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
alifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
context = public
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport = udp
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
alifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
srvlookup = yes
subscribecontext = default
accept_outofcall_message = yes
outofcall_message_context = astsms
allowsubscribe = yes
subscribecontext = default
qualifyfreq = 600
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
register = 620:[email protected]:5060/620
register = 621:[email protected]:5060/621
register = 622:[email protected]:5060/622
register = 623:[email protected]:5060/623
register = 624:[email protected]:5060/624
register = 625:[email protected]:5060/625
register = 626:[email protected]:5060/626
register = 627:[email protected]:5060/627
register = 628:[email protected]:5060/628
callgroup = 1,1-100
pickupgroup = 1

; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
; Specifying a port in a SIP peer definition or
; when dialing outbound calls will supress SRV
; lookups for that peer or call.

;pedantic=yes                   ; Enable checking of tags in headers,
; international character conversions in URIs
; and multiline formatted headers for strict
; SIP compatibility (defaults to "yes")

; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
;tos_sip=cs3                    ; Sets TOS for SIP packets.
;tos_audio=ef                   ; Sets TOS for RTP audio packets.
;tos_video=af41                 ; Sets TOS for RTP video packets.
;tos_text=af41                  ; Sets TOS for RTP text packets.

;cos_sip=3                      ; Sets 802.1p priority for SIP packets.
;cos_audio=5                    ; Sets 802.1p priority for RTP audio packets.
;cos_video=4                    ; Sets 802.1p priority for RTP video packets.
;cos_text=3                     ; Sets 802.1p priority for RTP text packets.


;maxexpiry=3600                 ; Maximum allowed time of incoming registrations (seconds)
;minexpiry=60                   ; Minimum length of registrations (default 60)
;defaultexpiry=120              ; Default length of incoming/outgoing registration
;submaxexpiry=3600              ; Maximum allowed time of incoming subscriptions (seconds), default: maxexpiry
;subminexpiry=60                ; Minimum length of subscriptions, default: minexpiry
;mwiexpiry=3600                 ; Expiry time for outgoing MWI subscriptions
;maxforwards=70            ; Setting for the SIP Max-Forwards: header (loop prevention)
; Default value is 70
;qualifyfreq=60                 ; Qualification: How often to check for the host to be up in seconds
; and reported in milliseconds with sip show settings.
; Set to low value if you use low timeout for NAT of UDP sessions
; Default: 60
;qualifygap=100            ; Number of milliseconds between each group of peers being qualified
; Default: 100
;qualifypeers=1            ; Number of peers in a group to be qualified at the same time
; Default: 1
;keepalive=60                   ; Interval at which keepalive packets should be sent to a peer
; Valid options are yes (60 seconds), no, or the number of seconds.
; Default: 0
;notifymimetype=text/plain      ; Allow overriding of mime type in MWI NOTIFY
;buggymwi=no                    ; Cisco SIP firmware doesn't support the MWI RFC
; fully. Enable this option to not get error messages
; when sending MWI to phones with this bug.
;mwi_from=asterisk              ; When sending MWI NOTIFY requests, use this setting in
; the From: header as the "name" portion. Also fill the
; "user" portion of the URI in the From: header with this
; value if no fromuser is set
; Default: empty
;vmexten=voicemail              ; dialplan extension to reach mailbox sets the
; Message-Account in the MWI notify message
; defaults to "asterisk"

; Codec negotiation
;
; When Asterisk is receiving a call, the codec will initially be set to the
; first codec in the allowed codecs defined for the user receiving the call
; that the caller also indicates that it supports. But, after the caller
; starts sending RTP, Asterisk will switch to using whatever codec the caller
; is sending.
;
; When Asterisk is placing a call, the codec used will be the first codec in
; the allowed codecs that the callee indicates that it supports. Asterisk will
; *not* switch to whatever codec the callee is sending.
;
;preferred_codec_only=yes       ; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.

;disallow=all                   ; First disallow all codecs
;allow=ulaw                     ; Allow codecs in order of preference
;allow=ilbc                     ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
; for framing options
;autoframing=yes        ; Set packetization based on the remote endpoint's (ptime)
; preferences. Defaults to no.
;
; 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.
;
; This option may be specified globally, or on a per-user or per-peer basis.
;
;mohinterpret=default
;
; This 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
;
;parkinglot=plaza               ; Sets the default parking lot for call parking
; This may also be set for individual users/peers
; Parkinglots are configured in features.conf
;language=en                    ; Default language setting for all users/peers
; This may also be set for individual users/peers
;tonezone=se            ; Default tonezone for all users/peers
; This may also be set for individual users/peers

;relaxdtmf=yes                  ; Relax dtmf handling
;trustrpid = no                 ; If Remote-Party-ID should be trusted
;sendrpid = yes                 ; If Remote-Party-ID should be sent (defaults to no)
;sendrpid = rpid                ; Use the "Remote-Party-ID" header
; to send the identity of the remote party
; This is identical to sendrpid=yes
;sendrpid = pai                 ; Use the "P-Asserted-Identity" header
; to send the identity of the remote party
;rpid_update = no               ; In certain cases, the only method by which a connected line
; change may be immediately transmitted is with a SIP UPDATE request.
; If communicating with another Asterisk server, and you wish to be able
; transmit such UPDATE messages to it, then you must enable this option.
; Otherwise, we will have to wait until we can send a reinvite to
; transmit the information.
;trust_id_outbound = no         ; Controls whether or not we trust this peer with private identity
; information (when the remote party has callingpres=prohib or equivalent).
; no - RPID/PAI headers will not be included for private peer information
; yes - RPID/PAI headers will include the private peer information. Privacy
;       requirements will be indicated in a Privacy header for sendrpid=pai
; legacy - RPID/PAI will be included for private peer information. In the
;       case of sendrpid=pai, private data that would be included in them
;       will be anonymized. For sendrpid=rpid, private data may be included
;       but the remote party's domain will be anonymized. The way legacy
;       behaves may violate RFC-3325, but it follows historic behavior.
; This option is set to 'legacy' by default
;prematuremedia=no              ; Some ISDN links send empty media frames before 
; the call is in ringing or progress state. The SIP 
; channel will then send 183 indicating early media
; which will be empty - thus users get no ring signal.
; Setting this to "yes" will stop any media before we have
; call progress (meaning the SIP channel will not send 183 Session
; Progress for early media). Default is "yes". Also make sure that
; the SIP peer is configured with progressinband=never. 
;
; In order for "noanswer" applications to work, you need to run
; the progress() application in the priority before the app.

;progressinband=never           ; If we should generate in-band ringing always
; use 'never' to never use in-band signalling, even in cases
; where some buggy devices might not render it
; Valid values: yes, no, never Default: never
;useragent=Asterisk PBX         ; Allows you to change the user agent string
; The default user agent string also contains the Asterisk
; version. If you don't want to expose this, change the
; useragent string.
;promiscredir = no              ; If yes, allows 302 or REDIR to non-local SIP address
; Note that promiscredir when redirects are made to the
; local system will cause loops since Asterisk is incapable
; of performing a "hairpin" call.
;usereqphone = no               ; If yes, ";user=phone" is added to uri that contains
; a valid phone number
;dtmfmode = rfc2833             ; Set default dtmfmode for sending DTMF. Default: rfc2833
; Other options:
; info : SIP INFO messages (application/dtmf-relay)
; shortinfo : SIP INFO messages (application/dtmf)
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
; auto : Use rfc2833 if offered, inband otherwise

;compactheaders = yes           ; send compact sip headers.
;
;videosupport=yes               ; Turn on support for SIP video. You need to turn this
; on in this section to get any video support at all.
; You can turn it off on a per peer basis if the general
; video support is enabled, but you can't enable it for
; one peer only without enabling in the general section.
; If you set videosupport to "always", then RTP ports will
; always be set up for video, even on clients that don't
; support it.  This assists callfile-derived calls and
; certain transferred calls to use always use video when
; available. [yes|NO|always]

;textsupport=no                 ; Support for ITU-T T.140 realtime text.
; The default value is "no".

;maxcallbitrate=384             ; Maximum bitrate for video calls (default 384 kb/s)
; Videosupport and maxcallbitrate is settable
; for peers and users as well
;authfailureevents=no           ; generate manager "peerstatus" events when peer can't
; authenticate with Asterisk. Peerstatus will be "rejected".
;alwaysauthreject = yes         ; When an incoming INVITE or REGISTER is to be rejected,
; for any reason, always reject with an identical response
; equivalent to valid username and invalid password/hash
; instead of letting the requester know whether there was
; a matching user or peer for their request.  This reduces
; the ability of an attacker to scan for valid SIP usernames.
; This option is set to "yes" by default.

;auth_options_requests = yes    ; Enabling this option will authenticate OPTIONS requests just like
; INVITE requests are.  By default this option is disabled.

;accept_outofcall_message = no  ; Disable this option to reject all MESSAGE requests outside of a
; call.  By default, this option is enabled.  When enabled, MESSAGE
; requests are passed in to the dialplan.

;outofcall_message_context = messages ; Context all out of dialog msgs are sent to. When this
; option is not set, the context used during peer matching
; is used. This option can be defined at both the peer and
; global level.

;auth_message_requests = yes    ; Enabling this option will authenticate MESSAGE requests.
; By default this option is enabled.  However, it can be disabled
; should an application desire to not load the Asterisk server with
; doing authentication and implement end to end security in the
; message body.

;g726nonstandard = yes          ; If the peer negotiates G726-32 audio, use AAL2 packing
; order instead of RFC3551 packing order (this is required
; for Sipura and Grandstream ATAs, among others). This is
; contrary to the RFC3551 specification, the peer _should_
; be negotiating AAL2-G726-32 instead :-(
;outboundproxy=proxy.provider.domain            ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain:8080       ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain,force      ; Send ALL outbound signalling to proxy, ignoring route: headers
;outboundproxy=tls://proxy.provider.domain      ; same as '=proxy.provider.domain' except we try to connect with tls
;outboundproxy=192.0.2.1                        ; IPv4 address literal (default port is 5060)
;outboundproxy=2001:db8::1                      ; IPv6 address literal (default port is 5060)
;outboundproxy=192.168.0.2.1:5062               ; IPv4 address literal with explicit port
;outboundproxy=[2001:db8::1]:5062               ; IPv6 address literal with explicit port
;                                               ; (could also be tcp,udp) - defining transports on the proxy line only
;                                               ; applies for the global proxy, otherwise use the transport= option

;supportpath=yes        ; This activates parsing and handling of Path header as defined in RFC 3327. This enables
; Asterisk to route outgoing out-of-dialog requests via a set of proxies by using a pre-loaded
; route-set defined by the Path headers in the REGISTER request.
; NOTE: There are multiple things to consider with this setting:
;  * As this influences routing of SIP requests make sure to not trust Path headers provided
;    by the user's SIP client (the proxy in front of Asterisk should remove existing user
;    provided Path headers).
;  * When a peer has both a path and outboundproxy set, the path will be added to Route: header
;    but routing to next hop is done using the outboundproxy.
;  * If set globally, not only will all peers use the Path header, but outbound REGISTER
;    requests from Asterisk will add path to the Supported header.

;rtsavepath=yes                 ; If using dynamic realtime, store the path headers

;matchexternaddrlocally = yes     ; Only substitute the externaddr or externhost setting if it matches
; your localnet setting. Unless you have some sort of strange network
; setup you will not need to enable this.

;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from registering
; as any IP address used for staticly defined
; hosts.  This helps avoid the configuration
; error of allowing your users to register at
; the same address as a SIP provider.

;contactdeny=0.0.0.0/0.0.0.0           ; Use contactpermit and contactdeny to
;contactpermit=172.16.0.0/255.255.0.0  ; restrict at what IPs your users may
; register their phones.
;contactacl=named_acl_example          ; Use named ACLs defined in acl.conf

;rtp_engine=asterisk            ; RTP engine to use when communicating with the device

;
; 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 and have a "regexten=" configuration item.
; Multiple contexts may be specified by separating them with '&'. The
; actual extension is the 'regexten' parameter of the registering peer or its
; name if 'regexten' is not provided.  If more than one context is provided,
; the context must be specified within regexten by appending the desired
; context after '@'.  More than one regexten may be supplied if they are
; separated by '&'.  Patterns may be used in regexten.
;
;regcontext=sipregistrations
;regextenonqualify=yes          ; Default "no"
; If you have qualify on and the peer becomes unreachable
; this setting will enforce inactivation of the regexten
; extension for the peer
;legacy_useroption_parsing=yes    ; Default "no"      ; If you have this option enabled and there are semicolons
; in the user field of a sip URI, the field be truncated
; at the first semicolon seen. This effectively makes
; semicolon a non-usable character for peer names, extensions,
; and maybe other, less tested things.  This can be useful
; for improving compatability with devices that like to use
; user options for whatever reason.  The behavior is similar to
; how SIP URI's were typically handled in 1.6.2, hence the name.

;send_diversion=no              ; Default "yes"     ; Asterisk normally sends Diversion headers with certain SIP
; invites to relay data about forwarded calls. If this option
; is disabled, Asterisk won't send Diversion headers unless
; they are added manually.

; 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


;use_q850_reason = no ; Default "no"
; Set to yes add Reason header and use Reason header if it is available.

; When the Transfer() application sends a REFER SIP message, extra headers specified in
; the dialplan by way of SIPAddHeader are sent out with that message. 1.8 and earlier did not
; add the extra headers. To revert to 1.8- behavior, call SIPRemoveHeader with no arguments
; before calling Transfer() to remove all additional headers from the channel. The setting
; below is for transitional compatibility only.
;
;refer_addheaders=yes    ; on by default

;autocreatepeer=no             ; Allow any UAC not explicitly defined to register
; WITHOUT AUTHENTICATION. Enabling this options poses a high
; potential security risk and should be avoided unless the
; server is behind a trusted firewall.
; If set to "yes", then peers created in this fashion
; are purged during SIP reloads.
; When set to "persist", the peers created in this fashion
; are not purged during SIP reloads.

;
;------------------------ TLS settings ------------------------------------------------------------
;tlscertfile=</path/to/certificate.pem> ; Certificate chain (*.pem format only) to use for TLS connections
; The certificates must be sorted starting with the subject's certificate
; and followed by intermediate CA certificates if applicable.
; Default is to look for "asterisk.pem" in current directory

;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections.
; If no tlsprivatekey is specified, tlscertfile is searched for
; for both public and private key.

;tlscafile=</path/to/certificate>
;        If the server your connecting to uses a self signed certificate
;        you should have their certificate installed here so the code can
;        verify the authenticity of their certificate.

;tlscapath=</path/to/ca/dir>
;        A directory full of CA certificates.  The files must be named with
;        the CA subject name hash value.
;        (see man SSL_CTX_load_verify_locations for more info)

;tlsdontverifyserver=[yes|no]
;        If set to yes, don't verify the servers certificate when acting as
;        a client.  If you don't have the server's CA certificate you can
;        set this and it will connect without requiring tlscafile to be set.
;        Default is no.

;tlscipher=<SSL cipher string>
;        A string specifying which SSL ciphers to use or not use
;        A list of valid SSL cipher strings can be found at:
;                http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
;
;tlsclientmethod=tlsv1     ; values include tlsv1, sslv3, sslv2.
; Specify protocol for outbound client connections.
; If left unspecified, the default is sslv2.
;
;--------------------------- SIP timers ----------------------------------------------------
; These timers are used primarily in INVITE transactions.
; The default for Timer T1 is 500 ms or the measured run-trip time between
; Asterisk and the device if you have qualify=yes for the device.
;
;t1min=100                      ; Minimum roundtrip time for messages to monitored hosts
; Defaults to 100 ms
;timert1=500                    ; Default T1 timer
; Defaults to 500 ms or the measured round-trip
; time to a peer (qualify=yes).
;timerb=32000                   ; Call setup timer. If a provisional response is not received
; in this amount of time, the call will autocongest
; Defaults to 64*timert1

;--------------------------- RTP timers ----------------------------------------------------
; These timers are currently used for both audio and video streams. The RTP timeouts
; are only applied to the audio channel.
; The settings are settable in the global section as well as per device
;
;rtptimeout=60                  ; Terminate call if 60 seconds of no RTP or RTCP activity
; on the audio channel
; when we're not on hold. This is to be able to hangup
; a call in the case of a phone disappearing from the net,
; like a powerloss or grandma tripping over a cable.
;rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP or RTCP activity
; on the audio channel
; when we're on hold (must be > rtptimeout)
;rtpkeepalive=<secs>            ; Send keepalives in the RTP stream to keep NAT open
; (default is off - zero)

;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
; This mechanism can detect and reclaim SIP channels that do not terminate through normal
; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
; The operation of Session-Timers is driven by the following configuration parameters:
;
; * session-timers    - Session-Timers feature operates in the following three modes:
;                            originate : Request and run session-timers always
;                            accept    : Run session-timers only when requested by other UA
;                            refuse    : Do not run session timers in any case
;                       The default mode of operation is 'accept'.
; * session-expires   - Maximum session refresh interval in seconds. Defaults to 1800 secs.
; * session-minse     - Minimum session refresh interval in seconds. Defualts to 90 secs.
; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
;                            uac - Default to the caller initially refreshing when possible
;                            uas - Default to the callee initially refreshing when possible
;
; Note that, due to recommendations in RFC 4028, Asterisk will always honor the other
; endpoint's preference for who will handle refreshes. Asterisk will never override the
; preferences of the other endpoint. Doing so could result in Asterisk and the endpoint
; fighting over who sends the refreshes. This holds true for the initiation of session
; timers and subsequent re-INVITE requests whether Asterisk is the caller or callee, or
; whether Asterisk is currently the refresher or not.
;
;session-timers=originate
;session-expires=600
;session-minse=90
;session-refresher=uac
;
;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes                 ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration.
; NOTE: You cannot use the CLI to turn it off. You'll
; need to edit this and reload the config.
;recordhistory=yes              ; Record SIP history by default
; (see sip history / sip no history)
;dumphistory=yes                ; Dump SIP history at end of SIP dialogue
; SIP history is output to the DEBUG logging channel


;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
; You can subscribe to the status of extensions with a "hint" priority
; (See extensions.conf.sample for examples)
; chan_sip support two major formats for notifications: dialog-info and SIMPLE
;
; You will get more detailed reports (busy etc) if you have a call counter enabled
; for a device.
;
; If you set the busylevel, we will indicate busy when we have a number of calls that
; matches the busylevel treshold.
;
; For queues, you will need this level of detail in status reporting, regardless
; if you use SIP subscriptions. Queues and manager use the same internal interface
; for reading status information.
;
; Note: Subscriptions does not work if you have a realtime dialplan and use the
; realtime switch.
;
;allowsubscribe=no              ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default     ; Set a specific context for SUBSCRIBE requests
; Useful to limit subscriptions to local extensions
; Settable per peer/user also
;notifyringing = no             ; Control whether subscriptions already INUSE get sent
; RINGING when another call is sent (default: yes)
;notifyhold = yes               ; Notify subscriptions on HOLD state (default: no)
; Turning on notifyringing and notifyhold will add a lot
; more database transactions if you are using realtime.
;notifycid = yes                ; Control whether caller ID information is sent along with
; dialog-info+xml notifications (supported by snom phones).
; Note that this feature will only work properly when the
; incoming call is using the same extension and context that
; is being used as the hint for the called extension.  This means
; that it won't work when using subscribecontext for your sip
; user or peer (if subscribecontext is different than context).
; This is also limited to a single caller, meaning that if an
; extension is ringing because multiple calls are incoming,
; only one will be used as the source of caller ID.  Specify
; 'ignore-context' to ignore the called context when looking
; for the caller's channel.  The default value is 'no.' Setting
; notifycid to 'ignore-context' also causes call-pickups attempted
; via SNOM's NOTIFY mechanism to set the context for the call pickup
; to PICKUPMARK.
;callcounter = yes              ; Enable call counters on devices. This can be set per
; device too.

;----------------------------------------- T.38 FAX SUPPORT ----------------------------------
;
; This setting is available in the [general] section as well as in device configurations.
; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
;
; t38pt_udptl = yes            ; Enables T.38 with FEC error correction.
; t38pt_udptl = yes,fec        ; Enables T.38 with FEC error correction.
; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
; t38pt_udptl = yes,none       ; Enables T.38 with no error correction.
;
; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that
; is based on an incorrect interpretation of the T.38 recommendation, and results in failures
; because Asterisk does not believe it can send T.38 packets of a reasonable size to that
; endpoint (Cisco media gateways are one example of this situation). In these cases, during a
; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL
; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you
; can set an override (globally, or on a per-device basis) to make Asterisk ignore the
; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead.
; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option,
; like this:
;
; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides
;                                       ; the other endpoint's provided value to assume we can
;                                       ; send 400 byte T.38 FAX packets to it.
;
; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists)
; based one or more events being detected. The events that can be detected are an incoming
; CNG tone or an incoming T.38 re-INVITE request.
;
; faxdetect = yes        ; Default 'no', 'yes' enables both CNG and T.38 detection
; faxdetect = cng        ; Enables only CNG detection
; faxdetect = t38        ; Enables only T.38 detection
;
;----------------------------------------- OUTBOUND SIP REGISTRATIONS  ------------------------
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
;       register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
;
;
;
; domain is either
;    - domain in DNS
;     - host name in DNS
;    - the name of a peer defined below or in realtime
; The domain is where you register your username, so your SIP uri you are registering to
; is username@domain
;
; If no extension is given, the 's' extension is used. The extension needs to
; be defined in extensions.conf to be able to accept calls from this SIP proxy
; (provider).
;
; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
; this is equivalent to having the following line in the general section:
;
;        register => username:secret@host/callbackextension
;
; and more readable because you don't have to write the parameters in two places
; (note that the "port" is ignored - this is a bug that should be fixed).
;
; Note that a register= line doesn't mean that we will match the incoming call in any
; other way than described above. If you want to control where the call enters your
; dialplan, which context, you want to define a peer with the hostname of the provider's
; server. If the provider has multiple servers to place calls to your system, you need
; a peer for each server.
;
; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
; contain a port number. Since the logical separator between a host and port number is a
; ':' character, and this character is already used to separate between the optional "secret"
; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
; they are blank. See the third example below for an illustration.
;
;
; Examples:
;
;register => 1234:[email protected]
;
;     This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@sip_proxy/1234
;
;    Register 2345 at sip provider 'sip_proxy'.  Calls from this provider
;    connect to local extension 1234 in extensions.conf, default context,
;    unless you configure a [sip_proxy] section below, and configure a
;    context.
;    Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
;    Tip 2: Use separate inbound and outbound sections for SIP providers
;           (instead of type=friend) if you have calls in both directions
;
;register => 3456@mydomain:5082::@mysipprovider.com
;
;    Note that in this example, the optional authuser and secret portions have
;    been left blank because we have specified a port in the user section
;
;register => tls://username:[email protected]
;
;    The 'transport' part defaults to 'udp' but may also be 'tcp', 'tls', 'ws', or 'wss'.
;    Using 'udp://' explicitly is also useful in case the username part
;    contains a '/' ('user/name').

;registertimeout=20             ; retry registration calls every 20 seconds (default)
;registerattempts=10            ; Number of registration attempts before we give up
; 0 = continue forever, hammering the other server
; until it accepts the registration
; Default is 0 tries, continue forever
;register_retry_403=yes         ; Treat 403 responses to registrations as if they were
; 401 responses and continue retrying according to normal
; retry rules.

;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
; by other phones. At this time, you can only subscribe using UDP as the transport.
; Format for the mwi register statement is:
;       mwi => user[:secret[:authuser]]@host[:port]/mailbox
;
; Examples:
;mwi => 1234:[email protected]/1234
;mwi => 1234:[email protected]:6969/1234
;mwi => 1234:password:[email protected]/1234
;mwi => 1234:password:[email protected]:6969/1234
;
; MWI received will be stored in the 1234 mailbox of the SIP_Remote context.
; It can be used by other phones by following the below:
; mailbox=1234@SIP_Remote
;----------------------------------------- NAT SUPPORT ------------------------
;
; WARNING: SIP operation behind a NAT is tricky and you really need
; to read and understand well the following section.
;
; When Asterisk is behind a NAT device, the "local" address (and port) that
; a socket is bound to has different values when seen from the inside or
; from the outside of the NATted network. Unfortunately this address must
; be communicated to the outside (e.g. in SIP and SDP messages), and in
; order to determine the correct value Asterisk needs to know:
;
; + whether it is talking to someone "inside" or "outside" of the NATted network.
;   This is configured by assigning the "localnet" parameter with a list
;   of network addresses that are considered "inside" of the NATted network.
;   IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
;   Multiple entries are allowed, e.g. a reasonable set is the following:
;
;      localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
;      localnet=10.0.0.0/255.0.0.0      ; Also RFC1918
;      localnet=172.16.0.0/12           ; Another RFC1918 with CIDR notation
;      localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
;
; + the "externally visible" address and port number to be used when talking
;   to a host outside the NAT. This information is derived by one of the
;   following (mutually exclusive) config file parameters:
;
;   a. "externaddr = hostname[:port]" specifies a static address[:port] to
;      be used in SIP and SDP messages.
;      The hostname is looked up only once, when [re]loading sip.conf .
;      If a port number is not present, use the port specified in the "udpbindaddr"
;      (which is not guaranteed to work correctly, because a NAT box might remap the
;      port number as well as the address).
;      This approach can be useful if you have a NAT device where you can
;      configure the mapping statically. Examples:
;
;        externaddr = 12.34.56.78          ; use this address.
;        externaddr = 12.34.56.78:9900     ; use this address and port.
;        externaddr = mynat.my.org:12600   ; Public address of my nat box.
;        externtcpport = 9900   ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT. 
;                               ; externtcpport will default to the externaddr or externhost port if either one is set. 
;        externtlsport = 12600  ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
;                               ; externtlsport port will default to the RFC designated port of 5061.    
;
;   b. "externhost = hostname[:port]" is similar to "externaddr" except
;      that the hostname is looked up every "externrefresh" seconds
;      (default 10s). This can be useful when your NAT device lets you choose
;      the port mapping, but the IP address is dynamic.
;      Beware, you might suffer from service disruption when the name server
;      resolution fails. Examples:
;
;        externhost=foo.dyndns.net       ; refreshed periodically
;        externrefresh=180               ; change the refresh interval
;
;   Note that at the moment all these mechanism work only for the SIP socket.
;   The IP address discovered with externaddr/externhost is reused for
;   media sessions as well, but the port numbers are not remapped so you
;   may still experience problems.
;
; NOTE 1: in some cases, NAT boxes will use different port numbers in
; the internal<->external mapping. In these cases, the "externaddr" and
; "externhost" might not help you configure addresses properly.
;
; NOTE 2: when using "externaddr" or "externhost", the address part is
; also used as the external address for media sessions. Thus, the port
; information in the SDP may be wrong!
;
; In addition to the above, Asterisk has an additional "nat" parameter to
; address NAT-related issues in incoming SIP or media sessions.
; In particular, depending on the 'nat= ' settings described below, Asterisk
; may override the address/port information specified in the SIP/SDP messages,
; and use the information (sender address) supplied by the network stack instead.
; However, this is only useful if the external traffic can reach us.
; The following settings are allowed (both globally and in individual sections):
;
;   nat = no                ; Do no special NAT handling other than RFC3581
;   nat = force_rport       ; Pretend there was an rport parameter even if there wasn't
;   nat = comedia           ; Send media to the port Asterisk received it from regardless
;                           ; of where the SDP says to send it.
;   nat = auto_force_rport  ; Set the force_rport option if Asterisk detects NAT (default)
;   nat = auto_comedia      ; Set the comedia option if Asterisk detects NAT
;
; The nat settings can be combined. For example, to set both force_rport and comedia
; one would set nat=force_rport,comedia. If any of the comma-separated options is 'no',
; Asterisk will ignore any other settings and set nat=no. If one of the "auto" settings
; is used in conjunction with its non-auto counterpart (nat=comedia,auto_comedia), then
; the non-auto option will be ignored.
;
; The RFC 3581-defined 'rport' parameter allows a client to request that Asterisk send
; SIP responses to it via the source IP and port from which the request originated
; instead of the address/port listed in the top-most Via header. This is useful if a
; client knows that it is behind a NAT and therefore cannot guess from what address/port
; its request will be sent. Asterisk will always honor the 'rport' parameter if it is
; sent. The force_rport setting causes Asterisk to always send responses back to the
; address/port from which it received requests; even if the other side doesn't support
; adding the 'rport' parameter.
;
; 'comedia RTP handling' refers to the technique of sending RTP to the port that the
; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is
; only partially related to RFC 4145 which was referred to as COMEDIA while it was in
; draft form. This method is used to accomodate endpoints that may be located behind
; NAT devices, and as such the address/port they tell Asterisk to send RTP packets to
; for their media streams is not the actual address/port that will be used on the nearer
; side of the NAT.
;
; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
; the nat setting in a peer definition, then the peer username will be discoverable
; by outside parties as Asterisk will respond to different ports for defined and
; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
; other, then valid peers with settings differing from those in the general section will
; be discoverable.
;
; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
; to receive them on.
;
; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
; the media_address configuration option. This is only applicable to the general section and
; can not be set per-user or per-peer.
;
; media_address = 172.16.42.1
;
; 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_sip 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
;
; ICE/STUN/TURN usage can be enabled globally or on a per-peer basis using the icesupport
; configuration option. When set to yes ICE support is enabled. When set to no it is disabled.
; It is disabled by default.
;
; icesupport = yes

;----------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
; no reason for Asterisk to stay in the media path, the media will be redirected.
; 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
; 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).
; The default setting is YES. If you have all clients
; behind a NAT, or for some other reason want Asterisk to
; stay in the audio path, you may want to turn this off.

; This setting also affect direct RTP
; at call setup (a new feature in 1.4 - setting up the
; call directly between the endpoints instead of sending
; a re-INVITE).

; Additionally this option does not disable all reINVITE operations.
; It only controls Asterisk generating reINVITEs for the specific
; purpose of setting up a direct media path. If a reINVITE is
; needed to switch a media stream to inactive (when placed on
; hold) or to T.38, it will still be done, regardless of this 
; setting. Note that direct T.38 is not supported.

;directmedia=nonat              ; An additional option is to allow media path redirection
; (reinvite) but only when the peer where the media is being
; sent is known to not be behind a NAT (as the RTP core can
; determine it based on the apparent IP address the media
; arrives from).

;directmedia=update             ; Yet a third option... use UPDATE for media path redirection,
; instead of INVITE. This can be combined with 'nonat', as
; 'directmedia=update,nonat'. It implies 'yes'.

;directmedia=outgoing           ; When sending directmedia reinvites, do not send an immediate
; reinvite on an incoming call leg. This option is useful when
; peered with another SIP user agent that is known to send
; immediate direct media reinvites upon call establishment. Setting
; 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
; 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
; callers INVITE. This will also fail if directmedia is enabled when
; the device is actually behind NAT.

;directmediadeny=0.0.0.0/0      ; Use directmediapermit and directmediadeny to restrict 
;directmediapermit=172.16.0.0/16; which peers should be able to pass directmedia to each other
; (There is no default setting, this is just an example)
; Use this if some of your phones are on IP addresses that
; can not reach each other directly. This way you can force 
; RTP to always flow through asterisk in such cases.
;directmediaacl=acl_example     ; Use named ACLs defined in acl.conf

;ignoresdpversion=yes           ; By default, Asterisk will honor the session version
; number in SDP packets and will only modify the SDP
; session if the version number changes. This option will
; force asterisk to ignore the SDP session version number
; and treat all SDP data as new data.  This is required
; for devices that send us non standard SDP packets
; (observed with Microsoft OCS). By default this option is
; off.

;sdpsession=Asterisk PBX        ; Allows you to change the SDP session name string, (s=)
; Like the useragent parameter, the default user agent string
; also contains the Asterisk version.
;sdpowner=root                  ; Allows you to change the username field in the SDP owner string, (o=)
; This field MUST NOT contain spaces
;encryption=no                  ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
; the peer does not support SRTP. Defaults to no.
;encryption_taglen=80           ; Set the auth tag length offered in the INVITE either 32/80 default 80
;
;avpf=yes                       ; Enable inter-operability with media streams using the AVPF RTP profile.
; This will cause all offers and answers to use AVPF (or SAVPF). This
; option may be specified at the global or peer scope.
;force_avp=yes            ; Force 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', and 'RTP/SAVPF' to be used for
; media streams when appropriate, even if a DTLS stream is present.
;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
;
;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 SIP UA registers successfully, the ip address,
; the origination port, the registration period, and the username of
; the UA will be set to database via realtime.
; If not present, defaults to 'yes'. Note: realtime peers will
; probably not function across reloads in the way that you expect, if
; you turn this option off.
;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.

;ignoreregexpire=yes            ; Enabling this setting has two functions:
;
; For non-realtime peers, when their registration expires, the
; information will _not_ be removed from memory or the Asterisk database
; if you attempt to place a call to the peer, the existing information
; will be used in spite of it having expired
;
; For realtime peers, when the peer is retrieved from realtime storage,
; the registration information will be used regardless of whether
; it has expired or not; if it expires while the realtime peer
; is still in memory (due to caching or other reasons), the
; information will not be removed from realtime storage

;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
; domains, each of which can direct the call to a specific context if desired.
; By default, all domains are accepted and sent to the default context or the
; context associated with the user/peer placing the call.
; REGISTER to non-local domains will be automatically denied if a domain
; list is configured.
;
; Domains can be specified using:
; domain=<domain>[,<context>]
; Examples:
; domain=myasterisk.dom
; domain=customer.com,customer-context
;
; In addition, all the 'default' domains associated with a server should be
; added if incoming request filtering is desired.
; autodomain=yes
;
; To disallow requests for domains not serviced by this server:
; allowexternaldomains=no

;domain=mydomain.tld,mydomain-incoming
; Add domain and configure incoming context
; for external calls to this domain
;domain=1.2.3.4                 ; Add IP address as local domain
; You can have several "domain" settings
;allowexternaldomains=no        ; Disable INVITE and REFER to non-local domains
; Default is yes
;autodomain=yes                 ; Turn this on to have Asterisk add local host
; name and local IP to domain list.

; fromdomain=mydomain.tld       ; When making outbound SIP INVITEs to
; non-peers, use your primary domain "identity"
; for From: headers instead of just your IP
; address. This is to be polite and
; it may be a mandatory requirement for some
; destinations which do not have a prior
; account relationship with your server.

;------------------------------ Advice of Charge CONFIGURATION --------------------------
; snom_aoc_enabled = yes;     ; This options turns on and off support for sending AOC-D and
; AOC-E to snom endpoints.  This option can be used both in the
; peer and global scope.  The default for this option is off.


;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
; SIP 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 SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.

; jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIP
; 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 SIP
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) 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".

;-----------------------------------------------------------------------------------

[authentication]
; Global credentials for outbound calls, i.e. when a proxy challenges your
; Asterisk server for authentication. These credentials override
; any credentials in peer/register definition if realm is matched.
;
; This way, Asterisk can authenticate for outbound calls to other
; realms. We match realm on the proxy challenge and pick an set of
; credentials from this list
; Syntax:
;        auth = <user>:<secret>@<realm>
;        auth = <user>#<md5secret>@<realm>
; Example:
;auth=mark:[email protected]
;
; You may also add auth= statements to [peer] definitions
; Peer auth= override all other authentication settings if we match on realm

;------------------------------------------------------------------------------
; DEVICE CONFIGURATION
;
; SIP entities have a 'type' which determines their roles within Asterisk.
; * For entities with 'type=peer':
;   Peers handle both inbound and outbound calls and are matched by ip/port, so for
;   The case of incoming calls from the peer, the IP address must match in order for
;   The invitation to work. This means calls made from either direction won't work if
;   The peer is unregistered while host=dynamic or if the host is otherise not set to
;   the correct IP of the sender.
; * For entities with 'type=user':
;   Asterisk users handle inbound calls only (meaning they call Asterisk, Asterisk can't
;   call them) and are matched by their authorization information (authname and secret).
;   Asterisk doesn't rely on their IP and will accept calls regardless of the host setting
;   as long as the incoming SIP invite authorizes successfully.
; * For entities with 'type=friend':
;   Asterisk will create the entity as both a friend and a peer. Asterisk will accept
;   calls from friends like it would for users, requiring only that the authorization
;   matches rather than the IP address. Since it is also a peer, a friend entity can
;   be called as long as its IP is known to Asterisk. In the case of host=dynamic,
;   this means it is necessary for the entity to register before Asterisk can call it.
; 
; Use remotesecret for outbound authentication, and secret for authenticating
; inbound requests. For historical reasons, if no remotesecret is supplied for an
; outbound registration or call, the secret will be used. 
;
; For device names, we recommend using only a-z, numerics (0-9) and underscore
;
; For local phones, type=friend works most of the time
;
; If you have one-way audio, you probably have NAT problems.
; If Asterisk is on a public IP, and the phone is inside of a NAT device
; you will need to configure nat option for those phones.
; Also, turn on qualify=yes to keep the nat session open
;
; Configuration options available
; --------------------
; context
; callingpres
; permit
; deny
; secret
; md5secret
; remotesecret
; transport
; dtmfmode
; directmedia
; nat
; callgroup
; pickupgroup
; language
; allow
; disallow
; autoframing
; insecure
; trustrpid
; trust_id_outbound
; progressinband
; promiscredir
; useclientcode
; accountcode
; setvar
; callerid
; amaflags
; callcounter
; busylevel
; allowoverlap
; allowsubscribe
; allowtransfer
; ignoresdpversion
; subscribecontext
; template
; videosupport
; maxcallbitrate
; rfc2833compensate
; Note: app_voicemail mailboxes must be in the form of mailbox@context.
; mailbox
; session-timers
; session-expires
; session-minse
; session-refresher
; t38pt_usertpsource
; regexten
; fromdomain
; fromuser
; host
; port
; qualify
; keepalive
; defaultip
; defaultuser
; rtptimeout
; rtpholdtimeout
; sendrpid
; outboundproxy
; rfc2833compensate
; callbackextension
; timert1
; timerb
; qualifyfreq
; t38pt_usertpsource
; contactpermit         ; Limit what a host may register as (a neat trick
; contactdeny           ; is to register at the same IP as a SIP provider,
; contactacl            ; then call oneself, and get redirected to that
;                       ; same location).
; directmediapermit
; directmediadeny
; directmediaacl
; unsolicited_mailbox
; use_q850_reason
; maxforwards
; encryption
; description        ; Used to provide a description of the peer in console output
; dtlsenable
; dtlsverify
; dtlsrekey
; dtlscertfile
; dtlsprivatekey
; dtlscipher
; dtlscafile
; dtlscapath
; dtlssetup
; dtlsfingerprint
; ignore_requested_pref ; Ignore the requested codec and determine the preferred codec
;                        ; from the peer's configuration.
;

;------------------------------------------------------------------------------
; DTLS-SRTP CONFIGURATION
;
; DTLS-SRTP support is available if the underlying RTP engine in use supports it.
;
; dtlsenable = yes                   ; Enable or disable DTLS-SRTP support
; dtlsverify = yes                   ; Verify that provided peer certificate and fingerprint are valid
;                     ; A value of 'yes' will perform both certificate and fingerprint verification
;                     ; A value of 'no' will perform no certificate or fingerprint verification
;                     ; A value of 'fingerprint' will perform ONLY fingerprint verification
;                     ; A value of 'certificate' will perform ONLY certficiate verification
; dtlsrekey = 60                     ; Interval at which to renegotiate the TLS session and rekey the SRTP session
;                                    ; If this is not set or the value provided is 0 rekeying will be disabled
; dtlscertfile = file                ; Path to certificate file to present
; dtlsprivatekey = file              ; Path to private key for certificate file
; dtlscipher = <SSL cipher string>   ; Cipher to use for TLS negotiation
;                                    ; A list of valid SSL cipher strings can be found at:
;                                    ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
; dtlscafile = file                  ; Path to certificate authority certificate
; dtlscapath = path                  ; Path to a directory containing certificate authority certificates
; dtlssetup = actpass                ; Whether we are willing to accept connections, connect to the other party, or both.
;                                    ; Valid options are active (we want to connect to the other party), passive (we want to
;                                    ; accept connections only), and actpass (we will do both). This value will be used in
;                                    ; the outgoing SDP when offering and for incoming SDP offers when the remote party sends
;                                    ; actpass
; dtlsfingerprint = sha-1            ; The hash to use for the fingerprint in SDP (valid options are sha-1 and sha-256)

;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
; We match on IP address of the proxy for incoming calls
; since we can not match on username (caller id)
;type=peer
;context=from-fwd
;host=fwd.pulver.com

;[sip_proxy-out]
;type=peer                        ; we only want to call out, not be called
;remotesecret=guessit             ; Our password to their service
;defaultuser=yourusername         ; Authentication user for outbound proxies
;fromuser=yourusername            ; Many SIP providers require this!
;fromdomain=provider.sip.domain
;host=box.provider.com
;transport=udp,tcp                ; This sets the default transport type to udp for outgoing, and will
;                                 ; accept both tcp and udp. The default transport type is only used for
;                                 ; outbound messages until a Registration takes place.  During the
;                                 ; peer Registration the transport type may change to another supported
;                                 ; type if the peer requests so.

;usereqphone=yes                  ; This provider requires ";user=phone" on URI
;callcounter=yes                  ; Enable call counter
;busylevel=2                      ; Signal busy at 2 or more calls
;outboundproxy=proxy.provider.domain  ; send outbound signaling to this proxy, not directly to the peer
;port=80                          ; The port number we want to connect to on the remote side
; Also used as "defaultport" in combination with "defaultip" settings

;--- sample definition for a provider
;[provider1]
;type=peer
;host=sip.provider1.com
;fromuser=4015552299              ; how your provider knows you
;remotesecret=youwillneverguessit ; The password we use to authenticate to them
;secret=gissadetdu                ; The password they use to contact us
;callbackextension=123            ; Register with this server and require calls coming back to this extension
;transport=udp,tcp                ; This sets the transport type to udp for outgoing, and will
;                                 ;   accept both tcp and udp. Default is udp. The first transport
;                                 ;   listed will always be used for outgoing connections.
;unsolicited_mailbox=4015552299   ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
;                                 ;   message count will be stored in the configured virtual mailbox. It can be used
;                                 ;   by any device supporting MWI by specifying <configured value>@SIP_Remote as the
;                                 ;   mailbox.

;
; Because you might have a large number of similar sections, it is generally
; convenient to use templates for the common parameters, and add them
; the the various sections. Examples are below, and we can even leave
; the templates uncommented as they will not harm:

[basic-options](!); a template
dtmfmode = rfc2833
context = from-office
type = friend

[natted-phone](!,basic-options); another template inheriting basic-options
directmedia = no
host = dynamic

[public-phone](!,basic-options); another template inheriting basic-options
directmedia = yes

[my-codecs](!); a template for my preferred codecs
disallow = all
allow = ilbc
allow = g729
allow = gsm
allow = g723
allow = ulaw
; Or, more simply:
;allow=!all,ilbc,g729,gsm,g723,ulaw

[ulaw-phone](!); and another one for ulaw-only
disallow = all
allow = ulaw
; Again, more simply:
;allow=!all,ulaw

; and finally instantiate a few phones
;
; [2133](natted-phone,my-codecs)
;        secret = peekaboo
; [2134](natted-phone,ulaw-phone)
;        secret = not_very_secret
; [2136](public-phone,ulaw-phone)
;        secret = not_very_secret_either
; ...
;

; Standard configurations not using templates look like this:
;
;[grandstream1]
;type=friend
;context=from-sip                ; Where to start in the dialplan when this phone calls
;recordonfeature=dynamicfeature1 ; Feature to use when INFO with Record: on is received.
;recordofffeature=dynamicfeature2 ; Feature to use when INFO with Record: off is received.
;callerid=John Doe <1234>        ; Full caller ID, to override the phones config
; on incoming calls to Asterisk
;description=Courtesy Phone      ; Description of the peer. Shown when doing 'sip show peers'.
;host=192.168.0.23               ; we have a static but private IP address
; No registration allowed
;directmedia=yes                 ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info                   ; either RFC2833 or INFO for the BudgeTone
;call-limit=1                    ; permit only 1 outgoing call and 1 incoming call at a time
; from the phone to asterisk (deprecated)
; 1 for the explicit peer, 1 for the explicit user,
; remember that a friend equals 1 peer and 1 user in
; memory
; There is no combined call counter for a "friend"
; so there's currently no way in sip.conf to limit
; to one inbound or outbound call per phone. Use
; the group counters in the dial plan for that.
;
;mailbox=1234@default            ; mailbox 1234 in voicemail context "default"
;disallow=all                    ; need to disallow=all before we can use allow=
;allow=ulaw                      ; Note: In user sections the order of codecs
; listed with allow= does NOT matter!
;allow=alaw
;allow=g723.1                    ; Asterisk only supports g723.1 pass-thru!
;allow=g729                      ; Pass-thru only unless g729 license obtained
;callingpres=allowed_passed_screen ; Set caller ID presentation
; See function CALLERPRES documentation for possible
; values.

;[xlite1]
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
;type=friend
;regexten=1234                   ; When they register, create extension 1234
;callerid="Jane Smith" <5678>
;host=dynamic                    ; This device needs to register
;directmedia=no                  ; Typically set to NO if behind NAT
;disallow=all
;allow=gsm                       ; GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
;registertrying=yes              ; Send a 100 Trying when the device registers.

;[snom]
;type=friend                     ; Friends place calls and receive calls
;context=from-sip                ; Context for incoming calls from this user
;secret=blah
;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
;language=de                     ; Use German prompts for this user
;host=dynamic                    ; This peer register with us
;dtmfmode=inband                 ; Choices are inband, rfc2833, or info
;defaultip=192.168.0.59          ; IP used until peer registers
;mailbox=1234@context,2345@context ; Mailbox(-es) for message waiting indicator
;subscribemwi=yes                ; Only send notifications if this phone
; subscribes for mailbox notification
;vmexten=voicemail               ; dialplan extension to reach mailbox
; sets the Message-Account in the MWI notify message
; defaults to global vmexten which defaults to "asterisk"
;disallow=all
;allow=ulaw                      ; dtmfmode=inband only works with ulaw or alaw!


;[polycom]
;type=friend                     ; Friends place calls and receive calls
;context=from-sip                ; Context for incoming calls from this user
;secret=blahpoly
;host=dynamic                    ; This peer register with us
;dtmfmode=rfc2833                ; Choices are inband, rfc2833, or info
;defaultuser=polly               ; Username to use in INVITE until peer registers
;defaultip=192.168.40.123
; Normally you do NOT need to set this parameter
;disallow=all
;allow=ulaw                      ; dtmfmode=inband only works with ulaw or alaw!
;progressinband=no               ; Polycom phones don't work properly with "never"


;[pingtel]
;type=friend
;secret=blah
;host=dynamic
;insecure=port                   ; Allow matching of peer by IP address without
; matching port number
;insecure=invite                 ; Do not require authentication of incoming INVITEs
;insecure=port,invite            ; (both)
;qualify=1000                    ; Consider it down if it's 1 second to reply
; Helps with NAT session
; qualify=yes uses default value
;qualifyfreq=60                  ; Qualification: How often to check for the
; host to be up in seconds
; Set to low value if you use low timeout for
; NAT of UDP sessions
;
; Call group and Pickup group should be in the range from 0 to 63
;
;callgroup=1,3-4                 ; We are in caller groups 1,3,4
;pickupgroup=1,3-5               ; We can do call pick-p for call group 1,3,4,5
;namedcallgroup=engineering,sales,netgroup,protgroup ; We are in named call groups engineering,sales,netgroup,protgroup
;namedpickupgroup=sales          ; We can do call pick-p for named call group sales
;defaultip=192.168.0.60          ; IP address to use if peer has not registered
;deny=0.0.0.0/0.0.0.0            ; ACL: Control access to this account based on IP address
;permit=192.168.0.60/255.255.255.0
;permit=192.168.0.60/24          ; we can also use CIDR notation for subnet masks
;permit=2001:db8::/32            ; IPv6 ACLs can be specified if desired. IPv6 ACLs
; apply only to IPv6 addresses, and IPv4 ACLs apply
; only to IPv4 addresses.
;acl=named_acl_example           ; Use named ACLs defined in acl.conf

;[cisco1]
;type=friend
;secret=blah
;qualify=200                     ; Qualify peer is no more than 200ms away
;host=dynamic                    ; This device registers with us
;directmedia=no                  ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee.  Some devices do not
; support this (especially if one of them is
; behind a NAT).
;defaultip=192.168.0.4           ; IP address to use until registration
;defaultuser=goran               ; Username to use when calling this device before registration
; Normally you do NOT need to set this parameter
;setvar=CUSTID=5678              ; Channel variable to be set for all calls from or to this device
;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.

;[pre14-asterisk]
;type=friend
;secret=digium
;host=dynamic
;rfc2833compensate=yes          ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
; You must have this turned on or DTMF reception will work improperly.
;t38pt_usertpsource=yes         ; Use the source IP address of RTP as the destination IP address for UDPTL packets
; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
; external IP address of the remote device. If port forwarding is done at the client side
; then UDPTL will flow to the remote device.

EXTENSIONS.conf
Code:
;!
;! Automatically generated configuration file
;! Filename: extensions.conf (/var/packages/Asterisk/target/etc/asterisk/extensions.conf)
;! Generator: Manager
;! Creation Date: Mon Sep 12 09:49:01 2016
;!
; extensions.conf - the Asterisk dial plan
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "dialplan reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI

;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static = yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command "dialplan save" too
;
writeprotect = no
;
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
; or HANGUP depending on Asterisk's best guess. This is the default.
;
; If autofallthrough is not set, then if an extension runs out of
; things to do, Asterisk will wait for a new extension to be dialed
; (this is the original behavior of Asterisk 1.0 and earlier).
;
;autofallthrough=no
;
;
;
; If extenpatternmatchnew is set (true, yes, etc), then a new algorithm that uses
; a Trie to find the best matching pattern is used. In dialplans
; with more than about 20-40 extensions in a single context, this
; new algorithm can provide a noticeable speedup.
; With 50 extensions, the speedup is 1.32x
; with 88 extensions, the speedup is 2.23x
; with 138 extensions, the speedup is 3.44x
; with 238 extensions, the speedup is 5.8x
; with 438 extensions, the speedup is 10.4x
; With 1000 extensions, the speedup is ~25x
; with 10,000 extensions, the speedup is 374x
; Basically, the new algorithm provides a flat response
; time, no matter the number of extensions.
;
; By default, the old pattern matcher is used.
;
; ****This is a new feature! *********************
; The new pattern matcher is for the brave, the bold, and
; the desperate. If you have large dialplans (more than about 50 extensions
; in a context), and/or high call volume, you might consider setting
; this value to "yes" !!
; Please, if you try this out, and are forced to return to the
; old pattern matcher, please report your reasons in a bug report
; on https://issues.asterisk.org. We have made good progress in providing
; something compatible with the old matcher; help us finish the job!
;
; This value can be switched at runtime using the cli command "dialplan set extenpatternmatchnew true"
; or "dialplan set extenpatternmatchnew false", so you can experiment to your hearts content.
;
;extenpatternmatchnew=no
;
; If clearglobalvars is set, global variables will be cleared
; and reparsed on a dialplan reload, or Asterisk reload.
;
; If clearglobalvars is not set, then global variables will persist
; through reloads, and even if deleted from the extensions.conf or
; one of its included files, will remain set to the previous value.
;
; NOTE: A complication sets in, if you put your global variables into
; the AEL file, instead of the extensions.conf file. With clearglobalvars
; set, a "reload" will often leave the globals vars cleared, because it
; is not unusual to have extensions.conf (which will have no globals)
; load after the extensions.ael file (where the global vars are stored).
; So, with "reload" in this particular situation, first the AEL file will
; clear and then set all the global vars, then, later, when the extensions.conf
; file is loaded, the global vars are all cleared, and then not set, because
; they are not stored in the extensions.conf file.
;
clearglobalvars = no
;
; User context is where entries from users.conf are registered.  The
; default value is 'default'
;
;userscontext=default
;
; You can include other config files, use the #include command
; (without the ';'). Note that this is different from the "include" command
; that includes contexts within other contexts. The #include command works
; in all asterisk configuration files.
;#include "filename.conf"
;#include <filename.conf>
;#include filename.conf
;
; You can execute a program or script that produces config files, and they
; will be inserted where you insert the #exec command. The #exec command
; works on all asterisk configuration files.  However, you will need to
; activate them within asterisk.conf with the "execincludes" option.  They
; are otherwise considered a security risk.
;#exec /opt/bin/build-extra-contexts.sh
;#exec /opt/bin/build-extra-contexts.sh --foo="bar"
;#exec </opt/bin/build-extra-contexts.sh --foo="bar">
;#exec "/opt/bin/build-extra-contexts.sh --foo=\"bar\""
;

; The "Globals" category contains global variables that can be referenced
; in the dialplan with the GLOBAL dialplan function:
; ${GLOBAL(VARIABLE)}
; ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
; Unix/Linux environmental variables can be reached with the ENV dialplan
; function: ${ENV(VARIABLE)}
;
[globals]
CONSOLE = Console/dsp
IAXINFO = guest
TRUNK = DAHDI/G2
TRUNKMSD = 1
FEATURES = 
DIALOPTIONS = 
RINGTIME = 20
FOLLOWMEOPTIONS = 
PAGING_HEADER = Intercom
620 = SIP/620
CID_620 = 620
trunk_1 = SIP/trunk_1
CID_trunk_1 = 621


trunk_3 = SIP/trunk_3
CID_trunk_3 = 623
trunk_4 = SIP/trunk_4
CID_trunk_4 = 624
trunk_5 = SIP/trunk_5
CID_trunk_5 = 625
trunk_6 = SIP/trunk_6
CID_trunk_6 = 626
trunk_7 = SIP/trunk_7
CID_trunk_7 = 627
trunk_8 = SIP/trunk_8
trunk_2 = SIP/trunk_2
CID_trunk_2 = 622
trunk_9 = SIP/trunk_9
CID_trunk_9 = 620
trunk_10 = SIP/trunk_10
CID_trunk_10 = 621
trunk_11 = SIP/trunk_11
CID_trunk_11 = 622
trunk_12 = SIP/trunk_12
CID_trunk_12 = 623
CID_trunk_8 = 628
trunk_13 = SIP/trunk_13
CID_trunk_13 = 624
trunk_14 = SIP/trunk_14
CID_trunk_14 = 625
trunk_15 = SIP/trunk_15
CID_trunk_15 = 626
trunk_16 = SIP/trunk_16
CID_trunk_16 = 627
FOLLOWME_6017 = 0
trunk_17 = SIP/trunk_17
CID_trunk_17 = 629



;TRUNK=IAX2/user:pass@provider

;FREENUMDOMAIN=mydomain.com                     ; domain to send on outbound
; freenum calls (uses outbound-freenum
; context)

;
; WARNING WARNING WARNING WARNING
; If you load any other extension configuration engine, such as pbx_ael.so,
; your global variables may be overridden by that file.  Please take care to
; use only one location to set global variables, and you will likely save
; yourself a ton of grief.
; WARNING WARNING WARNING WARNING
;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal.  In patterns, some characters have special meanings:
;
;   X - any digit from 0-9
;   Z - any digit from 1-9
;   N - any digit from 2-9
;   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
;   . - wildcard, matches anything remaining (e.g. _9011. matches
;    anything starting with 9011 excluding 9011 itself)
;   ! - wildcard, causes the matching process to complete as soon as
;       it can unambiguously determine that no other matches are possible
;
; For example, the extension _NXXXXXX would match normal 7 digit dialings,
; while _1NXXNXXXXXX would represent an area code plus phone number
; preceded by a one.
;
; Each step of an extension is ordered by priority, which must always start
; with 1 to be considered a valid extension.  The priority "next" or "n" means
; the previous priority plus one, regardless of whether the previous priority
; was associated with the current extension or not.  The priority "same" or "s"
; means the same as the previously specified priority, again regardless of
; whether the previous entry was for the same extension.  Priorities may be
; immediately followed by a plus sign and another integer to add that amount
; (most useful with 's' or 'n').  Priorities may then also have an alias, or
; label, in parentheses after their name which can be used in goto situations.
;
; Contexts contain several lines, one for each step of each extension.  One may
; include another context in the current one as well, optionally with a date
; and time.  Included contexts are included in the order they are listed.
; Switches may also be included within a context.  The order of matching within
; a context is always exact extensions, pattern match extensions, includes, and
; switches.  Includes are always processed depth-first.  So for example, if you
; would like a switch "A" to match before context "B", simply put switch "A" in
; an included context "C", where "C" is included in your original context
; before "B".
;
;[context]
;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
;
; Timing list for includes is
;
;   <time range>,<days of week>,<days of month>,<months>[,<timezone>]
;
; Note that ranges may be specified to wrap around the ends.  Also, minutes are
; fine-grained only down to the closest even minute.
;
;include => daytime,9:00-17:00,mon-fri,*,*
;include => weekend,*,sat-sun,*,*
;include => weeknights,17:02-8:58,mon-fri,*,*
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon receipt
; of a particular pattern.  The most commonly used example is of course '9'
; like this:
;
;ignorepat => 9
;
; 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 SIP and MGCP, which 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).
;

;
; Sample entries for extensions.conf
;
;
[dundi-e164-canonical]
;include => stdexten
;
; List canonical entries here
;
;exten => 12564286000,1,Gosub(6000,stdexten(IAX2/foo))
;exten => 12564286000,n,Goto(default,s,1)    ; exited Voicemail
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})

[dundi-e164-customers]
;
; If you are an ITSP or Reseller, list your customers here.
;
;exten => _12564286000,1,Dial(SIP/customer1)
;exten => _12564286001,1,Dial(IAX2/customer2)

[dundi-e164-via-pstn]
;
; If you are freely delivering calls to the PSTN, list them here
;
;exten => _1256428XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256325XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325

[dundi-e164-local]
;
; Context to put your dundi IAX2 or SIP user in for
; full access
;
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn

[dundi-e164-switch]
;
; Just a wrapper for the switch
;
switch => DUNDi/e164

[dundi-e164-lookup]
;
; Locally to lookup, try looking for a local E.164 solution
; then try DUNDi if we don't have one.
;
include => dundi-e164-local
include => dundi-e164-switch
;
; DUNDi can also be implemented as a Macro instead of using
; the Local channel driver.
;
[macro-dundi-e164]
;
; ARG1 is the extension to Dial
;
; Extension "s" is not a wildcard extension that matches "anything".
; In macros, it is the start extension. In most other cases,
; you have to goto "s" to execute that extension.
;
; Note: In old versions of Asterisk the PBX in some cases defaulted to
; extension "s" when a given extension was wrong (like in AMI originate).
; This is no longer the case.
;
; For wildcard matches, see above - all pattern matches start with
; an underscore.
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup

;
; Here are the entries you need to participate in the IAXTEL
; call routing system.  Most IAXTEL numbers begin with 1-700, but
; there are exceptions.  For more information, and to sign
; up, please go to www.gnophone.com or www.iaxtel.com
;
[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)

;
; The SWITCH statement permits a server to share the dialplan with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A -> B and B -> A), and the switched server needs
; to be on-line or else dialing can be severly delayed.
;
[iaxprovider]
;switch => IAX2/user:[key]@myserver/mycontext

[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${FILTER(0-9,${EXTEN:${GLOBAL(TRUNKMSD)}})})

[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint

[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld

[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider

;Include parkedcalls (or the context you define in features conf)
;to enable call parking.
include => parkedcalls
;
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote
; IAX switching you transparently get access to the remote
; Asterisk PBX
;
; switch => IAX2/user:password@bigserver/local
;
; An "lswitch" is like a switch but is literal, in that
; variable substitution is not performed at load time
; but is passed to the switch directly (presumably to
; be substituted in the switch routine itself)
;
; lswitch => Loopback/12${EXTEN}@othercontext
;
; An "eswitch" is like a switch but the evaluation of
; variable substitution is performed at runtime before
; being passed to the switch routine.
;
; eswitch => IAX2/context@${CURSERVER}

; The following two contexts are a template to enable the ability to dial
; ISN numbers. For more information about what an ISN number is, please see
; http://www.freenum.org.
;
; This is the dialing hook.  use:
; include => outbound-freenum

[outbound-freenum]
; We'll add more digits as needed. The purpose is to dial things
; like extension numbers at domains (ITAD number) so we're matching
; on lengths of 1 through 6 prior to the separator (the asterisk 
[*])
;
exten => _X*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)

[outbound-freenum2]
; This is the handler which performs the dialing logic. It is called
; from the [outbound-freenum] context
;
exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)})  ; make sure the suffix is all digits as well
same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1)
; filter out bad characters per the README-SERIOUSLY.best-practices.txt document
same => n,Set(TIMEOUT(absolute)=10800)
same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})  ; perform our lookup with freenum.org
same => n,GotoIf($["${isnresult}" != ""]?from)
same => n,Set(DIALSTATUS=CONGESTION)
same => n,Goto(fn-CONGESTION,1)
same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial)  ; check if we set the FREENUMDOMAIN global variable in [global]
same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})  ;    if we did set it, then we'll use it for our outbound dialing domain
same => n(dial),Dial(SIP/${isnresult},40)
same => n,Goto(fn-${DIALSTATUS},1)

exten => fn-BUSY,1,Busy()

exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same => n,Congestion()

[macro-trunkdial]
;
; Standard trunk dial macro (hangs up on a dialstatus that should
; terminate call)
;   ${ARG1} - What to dial
;
exten => s,1,Dial(${ARG1})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Hangup
exten => _s-.,1,NoOp

[stdexten]
;
; Standard extension subroutine:
;   ${EXTEN} - Extension
;   ${ARG1} - Device(s) to ring
;   ${ARG2} - Optional context in Voicemail
;
; Note that the current version will drop through to the next priority in the
; case of their pressing '#'.  This gives more flexibility in what do to next:
; you can prompt for a new extension, or drop the call, or send them to a
; general delivery mailbox, or...
;
; The use of the LOCAL() function is purely for convenience.  Any variable
; initially declared as LOCAL() will disappear when the innermost Gosub context
; in which it was declared returns.  Note also that you can declare a LOCAL()
; variable on top of an existing variable, and its value will revert to its
; previous value (before being declared as LOCAL()) upon Return.
;
exten => _X.,50000(stdexten),NoOp(Start stdexten)
exten => _X.,n,Set(LOCAL(ext)=${EXTEN})
exten => _X.,n,Set(LOCAL(dev)=${ARG1})
exten => _X.,n,Set(LOCAL(cntx)=${ARG2})
exten => _X.,n,Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})})
exten => _X.,n,Dial(${dev},20)  ; Ring the interface, 20 seconds maximum
exten => _X.,n,Goto(stdexten-${DIALSTATUS},1)  ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => stdexten-NOANSWER,1,Voicemail(${mbx},u)  ; If unavailable, send to voicemail w/ unavail announce
exten => stdexten-NOANSWER,n,Return()  ; If they press #, return to start

exten => stdexten-BUSY,1,Voicemail(${mbx},b)  ; If busy, send to voicemail w/ busy announce
exten => stdexten-BUSY,n,Return()  ; If they press #, return to start

exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1)  ; Treat anything else as no answer

exten => a,1,VoicemailMain(${mbx})  ; If they press *, send the user into VoicemailMain
exten => a,n,Return()

[stdPrivacyexten]
;
; Standard extension subroutine:
;   ${ARG1} - Extension
;   ${ARG2} - Device(s) to ring
;   ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
;   ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
;   ${ARG5} - Context in voicemail (if empty, then "default")
;
; See above note in stdexten about priority handling on exit.
;
exten => _X.,60000(stdPrivacyexten),NoOp(Start stdPrivacyexten)
exten => _X.,n,Set(LOCAL(ext)=${ARG1})
exten => _X.,n,Set(LOCAL(dev)=${ARG2})
exten => _X.,n,Set(LOCAL(dontcntx)=${ARG3})
exten => _X.,n,Set(LOCAL(tortcntx)=${ARG4})
exten => _X.,n,Set(LOCAL(cntx)=${ARG5})

exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
exten => _X.,n,Dial(${dev},20,p)  ; Ring the interface, 20 seconds maximum, call screening
; option (or use P for databased call _X.creening)
exten => _X.,n,Goto(stdexten-${DIALSTATUS},1)  ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => stdexten-NOANSWER,1,Voicemail(${mbx},u)  ; If unavailable, send to voicemail w/ unavail announce
exten => stdexten-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
exten => stdexten-NOANSWER,n,Return()  ; If they press #, return to start

exten => stdexten-BUSY,1,Voicemail(${mbx},b)  ; If busy, send to voicemail w/ busy announce
exten => stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
exten => stdexten-BUSY,n,Return()  ; If they press #, return to start

exten => stdexten-DONTCALL,1,Goto(${dontcntx},s,1)  ; Callee chose to send this call to a polite "Don't call again" script.

exten => stdexten-TORTURE,1,Goto(${tortcntx},s,1)  ; Callee chose to send this call to a telemarketer torture script.

exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1)  ; Treat anything else as no answer

exten => a,1,VoicemailMain(${mbx})  ; If they press *, send the user into VoicemailMain
exten => a,n,Return

[macro-page];
;
; Paging macro:
;
;       Check to see if SIP device is in use and DO NOT PAGE if they are
;
;   ${ARG1} - Device to page

exten => s,1,ChanIsAvail(${ARG1},s)  ; s is for ANY call
exten => s,n,GoToIf($[${AVAILSTATUS} = "1"]?autoanswer:fail)
exten => s,n(autoanswer),Set(_ALERT_INFO="RA")  ; This is for the PolyComs
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)  ; This is for the Grandstream, Snoms, and Others
exten => s,n,NoOp()  ; Add others here and Post on the Wiki!!!!
exten => s,n,Dial(${ARG1})
exten => s,n(fail),Hangup


[demo]
include => stdexten
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait(1)  ; Wait a second, just for fun
exten => s,n,Answer  ; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5)  ; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10)  ; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats)  ; Play a congratulatory message
exten => s,n(instruct),BackGround(demo-instruct)  ; Play some instructions
exten => s,n,WaitExten  ; Wait for an extension to be dialed.

exten => 2,1,BackGround(demo-moreinfo)  ; Give some more information.
exten => 2,n,Goto(s,instruct)

exten => 3,1,Set(CHANNEL(language)=fr)  ; Set language to french
exten => 3,n,Goto(s,restart)  ; Start with the congratulations

exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has
; voicemail, etc.
;
exten => 1234,1,Playback(transfer,skip)  ; "Please hold while..."
; (but skip if channel is not up)
exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
exten => 1234,n,Goto(default,s,1)  ; exited Voicemail

exten => 1235,1,Voicemail(1234,u)  ; Right to voicemail

exten => 1236,1,Dial(Console/dsp)  ; Ring forever
exten => 1236,n,Voicemail(1234,b)  ; Unless busy

;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks)  ; "Thanks for trying the demo"
exten => #,n,Hangup  ; Hang them up.

;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1)  ; If they take too long, give up
exten => i,1,Playback(invalid)  ; "That's not valid, try again"

;
; Create an extension, 500, for dialing the
; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry)  ; Let them know what's going on
exten => 500,n,Dial(IAX2/[email protected]/s@default)  ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo)  ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6)  ; Return to the start over message.

;
; Create an extension, 600, for evaluating echo latency.
;
exten => 600,1,Playback(demo-echotest)  ; Let them know what's going on
exten => 600,n,Echo  ; Do the echo test
exten => 600,n,Playback(demo-echodone)  ; Let them know it's over
exten => 600,n,Goto(s,6)  ; Start over

;
;    You can use the Macro Page to intercom a individual user
exten => 76245,1,Macro(page,SIP/Grandstream1)
; or if your peernames are the same as extensions
exten => _7XXX,1,Macro(page,SIP/${EXTEN})
;
;
; System Wide Page at extension 7999
;
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d)

; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
;
; Here's what a phone entry would look like (IXJ for example)
;
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,n,Goto(s,5)

;
;    The page context calls up the page macro that sets variables needed for auto-answer
;    It is in is own context to make calling it from the Page() application as simple as
;    Local/{peername}@page
;
[page]
exten => _X.,1,Macro(page,SIP/${EXTEN})

;[mainmenu]
;
; Example "main menu" context with submenu
;
;exten => s,1,Answer
;exten => s,n,Background(thanks)        ; "Thanks for calling press 1 for sales, 2 for support, ..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing                    ; Make them comfortable with 2 seconds of ringback
;exten => s,n,Wait,2
;exten => s,n,Background(submenuopts)    ; "Thanks for calling the sales department.  Press 1 for steve, 2 for..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)

[public]
;
; ATTENTION: If your Asterisk is connected to the internet and you do
; not have allowguest=no in sip.conf, everybody out there may use your
; public context without authentication.  In that case you want to
; double check which services you offer to the world.
;
include => demo

[default]
exten = 6000,1,Dial(SIP/6000)
exten = 6000,hint,SIP/6000,CustomPresence:6000
exten = 6001,1,Dial(SIP/6001)
exten = 6001,hint,SIP/6001,CustomPresence:6001
exten = 6002,1,Dial(SIP/6002)
exten = 6002,hint,SIP/6002,CustomPresence:6002
exten = 6003,1,Dial(SIP/6003)
exten = 6003,hint,SIP/6003,CustomPresence:6003
exten = 6004,1,Dial(SIP/6004)
exten = 6004,hint,SIP/6004,CustomPresence:6004
exten = 6005,1,Dial(SIP/6005)
exten = 6005,hint,SIP/6005,CustomPresence:6005
exten = 6006,1,Dial(SIP/6006)
exten = 6006,hint,SIP/6006,CustomPresence:6006
exten = 6007,1,Dial(SIP/6007)
exten = 6007,hint,SIP/6007,CustomPresence:6007
exten = 6008,1,Dial(SIP/6008)
exten = 6008,hint,SIP/6008,CustomPresence:6008
exten = 6009,1,Dial(SIP/6009)
exten = 6009,hint,SIP/6009,CustomPresence:6009
exten = 6010,1,Dial(SIP/6010)
exten = 6010,hint,SIP/6010,CustomPresence:6010
exten = 6011,1,Dial(SIP/6011)
exten = 6011,hint,SIP/6011,CustomPresence:6011
exten = 6012,1,Dial(SIP/6012)
exten = 6012,hint,SIP/6012,CustomPresence:6012
exten = 6013,1,Dial(SIP/6013)
exten = 6013,hint,SIP/6013,CustomPresence:6013
exten = 6014,1,Dial(SIP/6014)
exten = 6014,hint,SIP/6014,CustomPresence:6014
exten = _#6XXX,1,Set(MBOX=${EXTEN:1}@default)
exten = _#6XXX,n,VoiceMail(${MBOX})
exten = a,1,VoicemailMain(${MBOX})
exten = 99,1,VoiceMailMain(${CALLERID(num)}@default)




;include = demo \; This line was commented by ASTERISK GUI

;
; An extension like the one below can be used for FWD, Nikotel, sipgate etc.
; Note that you must have a [sipprovider] section in sip.conf
;
;exten => _41X.,1,Dial(SIP/${FILTER(0-9,${EXTEN:2})}@sipprovider,,r)

; Real extensions would go here. Generally you want real extensions to be
; 4 or 5 digits long (although there is no such requirement) and start with a
; single digit that is fairly large (like 6 or 7) so that you have plenty of
; room to overlap extensions and menu options without conflict.  You can alias
; them with names, too, and use global variables

;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1(Joe Schmoe) ; Channel hints for presence
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt)    ; permit transfer
;exten => 6245,n(dial),Dial(${HINT},20,rtT)    ; Use hint as listed
;exten => 6245,n,Voicemail(6245,u)        ; Voicemail (unavailable)
;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/[email protected])
;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
;exten => 6391,1,Dial(JINGLE/[email protected]/[email protected]) ;Dial via jingle using asterisk as the transport and calling mogorman.
;exten => 6394,1,Dial(Local/6275/n)        ; this will dial ${MARK}

;exten => 6275,1,Gosub(${EXTEN},stdexten(${MARK}))
; assuming ${MARK} is something like DAHDI/2
;exten => 6275,n,Goto(default,s,1)        ; exited Voicemail
;exten => mark,1,Goto(6275,1)            ; alias mark to 6275
;exten => 6536,1,Gosub(${EXTEN},stdexten(${WIL}))
; Ditto for wil
;exten => 6536,n,Goto(default,s,1)        ; exited Voicemail
;exten => wil,1,Goto(6236,1)

;If you want to subscribe to the status of a parking space, this is
;how you do it. Subscribe to extension 6600 in sip, and you will see
;the status of the first parking lot with this extensions' help
;exten => 6600,hint,park:701@parkedcalls
;exten => 6600,1,noop
;
;To subscribe to the availability of a free member in the 'markq' queue.
;Note: '_avail' is added to the QueueName
;exten => 8501,hint,Queue:markq_avail
;exten => 8501,1,Queue(markq)
;
; You can also monitor the status of a queue by providing a hint for a
; particular queue name.
;exten => 8502,hint,Queue:markq
;exten => 8502,1,Queue(markq)
;
; Some other handy things are an extension for checking voicemail via
; voicemailmain
;
;exten => 8500,1,VoicemailMain
;exten => 8500,n,Hangup
;
; Or a conference room (you'll need to edit meetme.conf to enable this room)
;
;exten => 8600,1,Meetme(1234)
;
; Or playing an announcement to the called party, as soon it answers
;
;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;

; example of a compartmentalized company called "acme"
;
; this is the context that your incoming IAX/SIP trunk dumps you in...
;[acme-incoming]
;exten => s,1,Wait(1)
;exten => s,n,Answer()
;exten => s,n(menu),Playback(acme/vm-brief-menu)
;exten => s,n(exten),Background(vm-enter-num-to-call)
;exten => s,n,WaitExten(5)
;exten => s,n(goodbye),Playback(vm-goodbye)
;exten => s,n(end),Hangup()
;
;include  => acme-extens
;
;exten => i,1,Playback(vm-invalid)
;exten => i,n,Goto(s,exten)            ; optionally, transfer to operator
;
;exten => t,1,Goto(s,goodbye)
;
; this is the context our internal SIP hardphones use (see sip.conf)
;
;[acme-internal]
;exten => s,1,Answer()
;exten => s,n(exten),Background(vm-enter-num-to-call)
;exten => s,n,WaitExten(5)
;exten => s,n(goodbye),Playback(vm-goodbye)
;exten => s,n(end),Hangup()
;
;include => trunkint
;include => trunkld
;include => trunklocal
;
;include => acme-extens
;
; you can test what your system sounds like to outside callers by dialing this
;exten => 777,1,DISA(no-password,acme-incoming)
;
; grouping of acme's extensions... never used directly, always included.
;
;[acme-extens]
;include => stdexten
;exten => 111,1,Gosub(111,stdexten(SIP/pete_1,acme))
;exten => 111,n,Goto(s,exten)
;
;exten => 112,1,Gosub(112,stdexten(SIP/nancy_1,acme))
;exten => 112,n,Goto(s,end)
;
; end of acme example

;
; Time context: you can patch this in via the following.
;
; [acme-internal]
; ...
; exten => 777,1,Gosub(time)
; exten => 777,n,Hangup()
;
; ...
; include => time
;
; Note: if you're geographically spread out, you can have SIP extensions
; specify their own local timezone in sip.conf as:
;
; [boi]
; type=friend
; context=acme-internal
; callerid="Boise Ofc. <2083451111>"
; ...
; ; use system-wide default timezone of MST7MDT
;
; [lws]
; type=friend
; context=acme-internal
; callerid="Lewiston Ofc. <2087431111>"
; ...
; setvar=timezone=PST8PDT
;
; "timezone" isn't a 'reserved' name in any way, and other places where
; the timezone is significant (e.g. calls to "SayUnixTime()", etc) will
; require modification as well.  Note that voicemail.conf already has
; a mechanism for timezones.
;

[time]
exten => _X.,30000(time),NoOp(Time: ${EXTEN} ${timezone})
exten => _X.,n,Wait(0.25)
exten => _X.,n,Answer()
; the amount of delay is set for English; you may need to adjust this time
; for other languages if there's no pause before the synchronizing beep.
exten => _X.,n,Set(FUTURETIME=$[${EPOCH} + 12])
exten => _X.,n,SayUnixTime(${FUTURETIME},Zulu,HNS)
exten => _X.,n,SayPhonetic(z)
; use the timezone associated with the extension (sip only), or system-wide
; default if one hasn't been set.
exten => _X.,n,SayUnixTime(${FUTURETIME},${timezone},HNS)
exten => _X.,n,Playback(spy-local)
exten => _X.,n,WaitUntil(${FUTURETIME})
exten => _X.,n,Playback(beep)
exten => _X.,n,Return()

;
; ANI context: use in the same way as "time" above
;

[ani]
exten => _X.,40000(ani),NoOp(ANI: ${EXTEN})
exten => _X.,n,Wait(0.25)
exten => _X.,n,Answer()
exten => _X.,n,Playback(vm-from)
exten => _X.,n,SayDigits(${CALLERID(ani)})
exten => _X.,n,Wait(1.25)
exten => _X.,n,SayDigits(${CALLERID(ani)})  ; playback again in case of missed digit
exten => _X.,n,Return()
; For more information on applications, just type "core show applications" at your
; friendly Asterisk CLI prompt.
;
; "core show application <command>" will show details of how you
; use that particular application in this file, the dial plan.
; "core show functions" will list all dialplan functions
; "core show function <COMMAND>" will show you more information about
; one function. Remember that function names are UPPER CASE.
[macro-stdexten]
exten = s,1,Set(__DYNAMIC_FEATURES=${FEATURES})
exten = s,2,Set(ORIG_ARG1=${ARG1})
exten = s,3,GotoIf($["${FOLLOWME_${ARG1}}" = "1"]?6:4)
exten = s,4,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
exten = s,5,Goto(s-${DIALSTATUS},1)
exten = s,6,Macro(stdexten-followme,${ARG1},${ARG2})
exten = s-NOANSWER,1,Voicemail(${ORIG_ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ORIG_ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ORIG_ARG1})
[macro-stdexten-followme]
exten = s,1,Answer
exten = s,2,Set(ORIG_ARG1=${ARG1})
exten = s,3,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
exten = s,4,Set(__FMCIDNUM=${CALLERID(num)})
exten = s,5,Set(__FMCIDNAME=${CALLERID(name)})
exten = s,6,Followme(${ORIG_ARG1},${FOLLOWMEOPTIONS})
exten = s,7,Voicemail(${ORIG_ARG1},u)
exten = s-NOANSWER,1,Voicemail(${ORIG_ARG1},u)
exten = s-BUSY,1,Voicemail(${ORIG_ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ORIG_ARG1})
[macro-pagingintercom]
exten = s,1,SIPAddHeader(Alert-Info: ${PAGING_HEADER})
exten = s,2,Page(${ARG1},${ARG2})
exten = s,3,Hangup
[conferences]
[ringgroups]

exten = 6402,1,Goto(ringroups-custom-3,s,1)
exten = 6403,1,Goto(ringroups-custom-4,s,1)



exten = 6405,1,Goto(ringroups-custom-6,s,1)




exten = 6407,1,Goto(ringroups-custom-8,s,1)
exten = 6409,1,Goto(ringroups-custom-10,s,1)

exten = 6411,1,Goto(ringroups-custom-12,s,1)



exten = 102,1,Goto(ringroups-custom-5,s,1)
exten = 104,1,Goto(ringroups-custom-7,s,1)




exten = 105,1,Goto(ringroups-custom-13,s,1)
exten = 101,1,Goto(ringroups-custom-1,s,1)

exten = 6400,1,Goto(ringroups-custom-14,s,1)

exten = 50,1,Goto(ringroups-custom-11,s,1)

exten = 21,1,Goto(ringroups-custom-2,s,1)
exten = 2,1,Goto(ringroups-custom-9,s,1)



[queues]
[voicemenus]
[voicemailgroups]
exten = 6600,1,NoOp(Samsys)
exten = 6600,2,VoiceMail(6001@default)
exten = 6601,1,NoOp(AWS)
exten = 6601,2,VoiceMail(6015@default)

[directory]
exten = #,1,Directory(default,default,f)


[page_an_extension]
[pagegroups]
[asterisk_guitools]
exten = executecommand,1,System(${command})
exten = executecommand,n,Hangup()
exten = record_vmenu,1,Answer
exten = record_vmenu,n,Playback(vm-intro)
exten = record_vmenu,n,Record(${var1},0,500,k)
exten = record_vmenu,n,Playback(vm-saved)
exten = record_vmenu,n,Playback(vm-goodbye)
exten = record_vmenu,n,Hangup
exten = play_file,1,Answer
exten = play_file,n,Playback(${var1})
exten = play_file,n,Hangup
[macro-local-callingrule-cid-0.1]
exten = s,1,Set(CALLERID(all)=${IF($[${LEN(${ARG4})} > 2]?${ARG4}:)})
exten = s,n,Goto(${ARG1},${ARG2},${ARG3})
[macro-trunkdial-failover-0.3]
exten = s,1,GotoIf($[${LEN(${FMCIDNUM})} > 6]?1-fmsetcid,1)
exten = s,n,GotoIf($[${LEN(${GLOBAL_OUTBOUNDCIDNAME})} > 1]?1-setgbobname,1)
exten = s,n,Set(CALLERID(num)=${IF($[${LEN(${CID_${CALLERID(num)}})} > 2]?${CID_${CALLERID(num)}}:)})
exten = s,n,Set(CALLERID(all)=${IF($[${LEN(${ARG5})} > 2]?${ARG5}:)})
exten = s,n,GotoIf($[${LEN(${CALLERID(num)})} > 6]?1-dial,1)
exten = s,n,Set(CALLERID(all)=${IF($[${LEN(${CID_${ARG3}})} > 6]?${CID_${ARG3}}:${GLOBAL_OUTBOUNDCID})})
exten = s,n,Set(CALLERID(all)=${IF($[${LEN(${ARG5})} > 2]?${ARG5}:)})
exten = s,n,Goto(1-dial,1)
exten = 1-setgbobname,1,Set(CALLERID(name)=${GLOBAL_OUTBOUNDCIDNAME})
exten = 1-setgbobname,n,Goto(s,3)
exten = 1-fmsetcid,1,Set(CALLERID(num)=${FMCIDNUM})
exten = 1-fmsetcid,n,Set(CALLERID(name)=${FMCIDNAME})
exten = 1-fmsetcid,n,Goto(s,4)
exten = 1-dial,1,Dial(${ARG1})
exten = 1-dial,n,Gotoif(${LEN(${ARG2})} > 0 ?1-${DIALSTATUS},1:1-out,1)
exten = 1-CHANUNAVAIL,1,Dial(${ARG2})
exten = 1-CHANUNAVAIL,n,Hangup()
exten = 1-CONGESTION,1,Dial(${ARG2})
exten = 1-CONGESTION,n,Hangup()
exten = 1-out,1,Hangup()
[queue-member-manager]
exten = handle_member,1,Verbose(2, Looping through queues to log in or out queue members)
exten = handle_member,n,Set(thisActiveMember=${CHANNEL(channeltype)}/${CHANNEL(peername)})
exten = handle_member,n,Set(queue_field=2)
exten = handle_member,n,Set(thisQueueXtn=${CUT(QUEUES,\,,${queue_field})})
exten = handle_member,n,While($[${EXISTS(${thisQueueXtn})}])
exten = handle_member,n,Macro(member-loginlogout)
exten = handle_member,n,Set(queue_field=$[${queue_field} + 1])
exten = handle_member,n,Set(thisQueueXtn=${CUT(QUEUES,\,,${queue_field})})
exten = handle_member,n,EndWhile()
[macro-member-loginlogout]
exten = s,1,Verbose(2, Logging queue member in or out of the request queue)
exten = s,n,Set(thisQueue=${thisQueueXtn})
exten = s,n,Set(queueMembers=${QUEUE_MEMBER_LIST(${thisQueue})})
exten = s,n,Set(field=1)
exten = s,n,Set(logged_in=0)
exten = s,n,Set(thisQueueMember=${CUT(queueMembers,\,,${field})})
exten = s,n,While($[${EXISTS(${thisQueueMember})}])
exten = s,n,GotoIf($["${thisQueueMember}" != "${thisActiveMember}"]?check_next)
exten = s,n,Set(logged_in=1)
exten = s,n,ExitWhile()
exten = s,n(check_next),Set(field=$[${field} + 1])
exten = s,n,Set(thisQueueMember=${CUT(queueMembers,\,,${field})})
exten = s,n,EndWhile()
exten = s,n,MacroIf($[${logged_in} = 0]?q_login:q_logout)
[macro-q_login]
exten = s,1,Verbose(2, Logging ${thisActiveMember} into the ${thisQueue} queue)
exten = s,n,AddQueueMember(${thisQueue},${thisActiveMember})
exten = s,n,Playback(silence/1)
exten = s,n,ExecIf($["${AQMSTATUS}" = "ADDED"]?Playback(agent-loginok):Playback(an-error-has-occurred))
[macro-q_logout]
exten = s,1,Verbose(2, Logged ${thisActiveMember} out of ${thisQueue} queue)
exten = s,n,RemoveQueueMember(${thisQueue},${thisActiveMember})
exten = s,n,Playback(silence/1)
exten = s,n,ExecIf($["${RQMSTATUS}" = "REMOVED"]?Playback(agent-loggedoff):Playback(an-error-has-occurred))
[DID_620]
include = DID_620_default

[CallingRule_outgoing]
exten = _0.,1,Macro(trunkdial-failover-0.3,${620}/${EXTEN:0},${trunk_9}/${EXTEN:0},620,trunk_9)

[CallingRule_outgoing2]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_8}/${EXTEN:0},${trunk_9}/${EXTEN:0},trunk_8,trunk_9)

[CallingRule_outgoing3]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_2}/${EXTEN:0},${trunk_9}/${EXTEN:0},trunk_2,trunk_9)



[astsms]
exten => _.,1,NoOp(SMS receiving dialplan invoked)
exten => _.,n,NoOp(To ${MESSAGE(to)})
exten => _.,n,NoOp(From ${MESSAGE(from)})
exten => _.,n,NoOp(Body ${MESSAGE(body)})
exten => _.,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)})
exten => _.,n,MessageSend(${ACTUALTO},${MESSAGE(from)})
exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
exten => _.,n,GotoIf($["${MESSAGE_SEND_STATUS}" != "SUCCESS"]?
exten => _.,n,Hangup()
exten => h,1,Hangup()
exten => _.,n(sendfailedmsg),Set(MESSAGE(body)="[${STRFTIME(${EPOCH},,
exten => _.,n,Set(ME_1=${CUT(MESSAGE(from),<,2)})
exten => _.,n,Set(ACTUALFROM=${CUT(ME_1,@,1)})
exten => _.,n,MessageSend(${ACTUALFROM},ServiceCenter)
exten => _.,n,Hangup()
exten => h,1,Hangup()




[DID_trunk_1]
include = DID_trunk_1_default
[DID_trunk_1_default]


[DID_trunk_3]
include = DID_trunk_3_default
[DID_trunk_3_default]


[DID_trunk_4]
include = DID_trunk_4_default
[DID_trunk_4_default]

[DID_trunk_5]
include = DID_trunk_5_default
[DID_trunk_5_default]
[DID_trunk_6]
include = DID_trunk_6_default
[DID_trunk_6_default]

[DID_trunk_7]
include = DID_trunk_7_default
[DID_trunk_7_default]
[DID_trunk_8]
include = DID_trunk_8_default
[DID_trunk_8_default]

[CallingRule_outgoing4]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_3}/${EXTEN:0},${trunk_9}/${EXTEN:0},trunk_3,trunk_9)
[DLPN_9962471]
include = CallingRule_outgoing4
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
include = pickup

[CallingRule_outgoing5]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_4}/${EXTEN:0},${trunk_9}/${EXTEN:0},trunk_4,trunk_9)
[DLPN_9962472]
include = CallingRule_outgoing5
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
include = pickup

[ringroups-custom-3]
exten = s,1,NoOp(Sascha Weingaertner)
exten = s,n,Dial(SIP/6008&SIP/6011,20,${DIALOPTIONS}i)
exten = s,n,Goto(voicemailgroups,6601,1)
[ringroups-custom-4]
exten = s,1,NoOp(Horst Muehleck)
exten = s,n,Dial(SIP/6009&SIP/6012,20,${DIALOPTIONS}i)
exten = s,n,Goto(voicemailgroups,6601,1)
[DID_621]
include = DID_621_default



[DID_623]
include = DID_623_default
[DID_624]
include = DID_624_default
[DID_625]
include = DID_625_default
[DID_626]
include = DID_626_default
[DID_627]
include = DID_627_default
[DID_628]
include = DID_628_default
[DID_621_default]
exten = s,1,Goto(ringroups-custom-5,s,1)


[DID_623_default]
exten = s,1,Goto(ringroups-custom-3,s,1)
[DID_624_default]
exten = s,1,Goto(ringroups-custom-4,s,1)
[DID_625_default]
[DID_626_default]
[DID_627_default]
[DID_628_default]




[DID_620_default]
exten = 620,1,Goto(ringroups-custom-1,s,1)
exten = 623,1,Goto(ringroups-custom-13,s,1)
exten = 621,1,Goto(ringroups-custom-5,s,1)
exten = 622,1,Goto(ringroups-custom-7,s,1)



[DLPN_9962470]
include = CallingRule_outgoing3
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
include = pickup

[CallingRule_outgoing6]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${EXTEN:0},${trunk_9}/${EXTEN:0},trunk_1,trunk_9)
[DLPN_5487960]
include = CallingRule_outgoing6
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
include = pickup

[DLPN_9989950]
include = CallingRule_outgoing
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
include = pickup

[DLPN_9962467]
include = CallingRule_outgoing2
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
include = pickup

[DID_trunk_2]
include = DID_trunk_2_default
[DID_trunk_2_default]
[CallingRule_outgoing7]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_5}/${EXTEN:0},${trunk_9}/${EXTEN:0},trunk_5,trunk_9)
[DLPN_9962473]
include = CallingRule_outgoing7
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
include = pickup

[ringroups-custom-6]
exten = s,1,NoOp(Werner Braun)
exten = s,n,Dial(SIP/6005,20,${DIALOPTIONS}i)
exten = s,n,Goto(voicemailgroups,6601,1)

[CallingRule_outgoing8]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_6}/${EXTEN:0},${trunk_9}/${EXTEN:0},trunk_6,trunk_9)
[DLPN_9962474]
include = CallingRule_outgoing8
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
include = pickup

[DID_trunk_9]
include = DID_trunk_9_default
[DID_trunk_9_default]
exten = 620,1,Goto(ringroups-custom-11,s,1)
exten = 622,1,Goto(ringroups-custom-9,s,1)
exten = 625,1,Goto(ringroups-custom-2,s,1)

[CallingRule_outgoing9]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_9}/${EXTEN:0},${620}/${EXTEN:0},trunk_9,620,45350)
[DLPN_45350]
include = CallingRule_outgoing9
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
include = pickup

[DID_trunk_10]
include = DID_trunk_10_default
[DID_trunk_10_default]
[DID_trunk_11]
include = DID_trunk_11_default
[DID_trunk_11_default]



[DID_trunk_12]
include = DID_trunk_12_default
[DID_trunk_12_default]
[DID_trunk_13]
include = DID_trunk_13_default
[DID_trunk_13_default]
[DID_trunk_14]
include = DID_trunk_14_default
[DID_trunk_14_default]
[DID_trunk_15]
include = DID_trunk_15_default
[DID_trunk_15_default]
[DID_trunk_16]
include = DID_trunk_16_default
[DID_trunk_16_default]

[ringroups-custom-8]
exten = s,1,NoOp(Christian Schwank)
exten = s,n,Dial(SIP/6000,20,${DIALOPTIONS}i)
exten = s,n,Goto(voicemailgroups,6601,1)
[ringroups-custom-10]
exten = s,1,NoOp(Phillip Liversidge)
exten = s,n,Dial(SIP/6016,20,${DIALOPTIONS}i)
exten = s,n,Goto(voicemailgroups,6601,1)




[ringroups-custom-12]
exten = s,1,NoOp(Marion Zwilling)
exten = s,n,Dial(SIP/6018,20,${DIALOPTIONS}i)
exten = s,n,Goto(voicemailgroups,6601,1)
[CallingRule_outgoing10]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_10}/${EXTEN:0},${620}/${EXTEN:0},trunk_10,620)
[CallingRule_outgoing11]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_11}/${EXTEN:0},${620}/${EXTEN:0},trunk_11,620)
[CallingRule_outgoing12]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_12}/${EXTEN:0},${620}/${EXTEN:0},trunk_12,620)
[CallingRule_outgoing13]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_13}/${EXTEN:0},${620}/${EXTEN:0},trunk_13,620)
[CallingRule_outgoing14]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_14}/${EXTEN:0},${620}/${EXTEN:0},trunk_14,620)
[CallingRule_outgoing15]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_15}/${EXTEN:0},${620}/${EXTEN:0},trunk_15,620)
[CallingRule_outgoing16]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_16}/${EXTEN:0},${620}/${EXTEN:0},trunk_16,620)
[DLPN_9478180]
include = CallingRule_outgoing10
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
include = pickup

[DLPN_9478181]
include = CallingRule_outgoing11
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
include = pickup

[DLPN_9478182]
include = CallingRule_outgoing12
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
include = pickup

[DLPN_9478183]
include = CallingRule_outgoing13
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
include = pickup

[DLPN_9478184]
include = CallingRule_outgoing14
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
include = pickup

[DLPN_9478185]
include = CallingRule_outgoing15
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
include = pickup

[DLPN_9478186]
include = CallingRule_outgoing16
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
include = pickup

[ringroups-custom-5]
exten = s,1,NoOp(SAMSYS Dennis Schreiber)
exten = s,n,Dial(SIP/6002&SIP/6006&SIP/6013,20,${DIALOPTIONS}i)
exten = s,n,Goto(voicemailgroups,6600,1)
[ringroups-custom-7]
exten = s,1,NoOp(SAMSYS Katja Braun)
exten = s,n,Dial(SIP/6014,20,${DIALOPTIONS}i)
exten = s,n,Goto(voicemailgroups,6600,1)

[ringroups-custom-13]
exten = s,1,NoOp(SAMSYS Jo Braun)
exten = s,n,Dial(SIP/6001&SIP/6003&SIP/6004,20,${DIALOPTIONS}i)
exten = s,n,Goto(voicemailgroups,6600,1)
[ringroups-custom-1]
exten = s,1,NoOp(SAMSYS Zentrale)
exten = s,n,Dial(SIP/6001&SIP/6004&SIP/6003&SIP/6014&SIP/6013&SIP/6002&SIP/6006,20,${DIALOPTIONS}i)
exten = s,n,Goto(voicemailgroups,6600,1)




[ringroups-custom-14]
exten = s,1,NoOp(AWS Dagmar Schwahn)
exten = s,n,Dial(SIP/6017,20,${DIALOPTIONS}i)
exten = s,n,Goto(voicemailgroups,6601,1)

[DID_trunk_17]
include = DID_trunk_17_default
[DID_trunk_17_default]
[CallingRule_outgoing1]
exten = _0.,1,Macro(trunkdial-failover-0.3,${trunk_7}/${EXTEN:0},${trunk_9}/${EXTEN:0},trunk_7,trunk_9,9962475)
[ringroups-custom-11]
exten = s,1,NoOp(AWS Zentrale)
exten = s,n,Dial(SIP/6000&SIP/6017&SIP/6018&SIP/6015,20,${DIALOPTIONS}i)
exten = s,n,Goto(voicemailgroups,6601,1)




[ringroups-custom-2]
exten = s,1,NoOp(AWS Gerd Stuber)
exten = s,n,Dial(SIP/6010&SIP/6000,20,${DIALOPTIONS}i)
exten = s,n,Goto(voicemailgroups,6601,1)
[ringroups-custom-9]
exten = s,1,NoOp(AWS Hardy Schwank)
exten = s,n,Dial(SIP/6015,20,${DIALOPTIONS}i)
exten = s,n,Goto(voicemailgroups,6601,1)
[pickup]
exten  =>  _*8.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:2})
exten  =>  _*8.,n,Pickup(${EXTEN:2}@PICKUPMARK)

FEATURES.conf
Code:
;!
;! Automatically generated configuration file
;! Filename: features.conf (/var/packages/Asterisk/target/etc/asterisk/features.conf)
;! Generator: Manager
;! Creation Date: Mon Sep 12 09:25:17 2016
;!
;
; Sample Call Features (transfer, monitor/mixmonitor, etc) configuration
;

; Asterisk 12 Note - All parking lot configuration is now done in res_parking.conf

[general]
pickupexten = *8
pickupsound = beep
pickupfailsound = beeperr
;transferdigittimeout => 3      ; Number of seconds to wait between digits when transferring a call
; (default is 3 seconds)
;xfersound = beep               ; to indicate an attended transfer is complete
;xferfailsound = beeperr        ; to indicate a failed transfer
;pickupexten = *8               ; Configure the pickup extension. (default is *8)
;pickupsound = beep             ; to indicate a successful pickup (default: no sound)
;pickupfailsound = beeperr      ; to indicate that the pickup failed (default: no sound)
;featuredigittimeout = 1000     ; Max time (ms) between digits for
; feature activation  (default is 1000 ms)
;recordingfailsound = beeperr   ; indicates that a one-touch monitor or one-touch mixmonitor feature failed
; to be applied to the call. (default: no sound)
;atxfernoanswertimeout = 15     ; Timeout for answer on attended transfer default is 15 seconds.
;atxferdropcall = no            ; If someone does an attended transfer, then hangs up before the transfer
; target answers, then by default, the system will try to call back the
; person that did the transfer.  If this is set to "yes", the ringing
; transfer target is immediately transferred to the transferee.
;atxferloopdelay = 10           ; Number of seconds to sleep between retries (if atxferdropcall = no)
;atxfercallbackretries = 2      ; Number of times to attempt to send the call back to the transferer.
; By default, this is 2.
;transferdialattempts = 3       ; Number of times that a transferer may attempt to dial an extension before
; being kicked back to the original call.
;transferretrysound = "beep"    ; Sound to play when a transferer fails to dial a valid extension.
;transferinvalidsound = "beeperr" ; Sound to play when a transferer fails to dial a valid extension and is out of retries.


; Note that the DTMF features listed below only work when two channels have answered and are bridged together.
; They can not be used while the remote party is ringing or in progress. If you require this feature you can use
; chan_local in combination with Answer to accomplish it.

[featuremap]
;blindxfer => #1                ; Blind transfer  (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
;disconnect => *0               ; Disconnect  (default is *) -- Make sure to set the H and/or h option in the Dial() or Queue() app call!
;automon => *1                  ; One Touch Record a.k.a. Touch Monitor -- Make sure to set the W and/or w option in the Dial() or Queue() app call!
;atxfer => *2                   ; Attended transfer  -- Make sure to set the T and/or t option in the Dial() or Queue()  app call!
;parkcall => #72                ; Park call (one step parking)  -- Make sure to set the K and/or k option in the Dial() app call!
;automixmon => *3               ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call!

[applicationmap]
; Note that the DYNAMIC_FEATURES channel variable must be set to use the features
; defined here.  The value of DYNAMIC_FEATURES should be the names of the features
; to allow the channel to use separated by '#'.  For example:
;
;    Set(__DYNAMIC_FEATURES=myfeature1#myfeature2#myfeature3)
;
; (Note: The two leading underscores allow these feature settings to be set
;  on the outbound channels, as well.  Otherwise, only the original channel
;  will have access to these features.)
;
; The syntax for declaring a dynamic feature is any of the following:
;
;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[,MOH_Class]]
;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,"<AppArguments>"[,MOH_Class]]
;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>([<AppArguments>])[,MOH_Class]

;
;  FeatureName   -> This is the name of the feature used when setting the
;                   DYNAMIC_FEATURES variable to enable usage of this feature.
;  DTMF_sequence -> This is the key sequence used to activate this feature.
;  ActivateOn    -> This is the channel of the call that the application will be executed
;                   on. Valid values are "self" and "peer". "self" means run the
;                   application on the same channel that activated the feature. "peer"
;                   means run the application on the opposite channel from the one that
;                   has activated the feature.
;  ActivatedBy   -> ActivatedBy is no longer honored.  The feature is activated by which
;                   channel DYNAMIC_FEATURES includes the feature is on.  Use predial
;                   to set different values of DYNAMIC_FEATURES on the channels.
;                   Historic values are: "caller", "callee", and "both".
;  Application   -> This is the application to execute.
;  AppArguments  -> These are the arguments to be passed into the application.  If you need
;                   commas in your arguments, you should use either the second or third
;                   syntax, above.
;  MOH_Class     -> This is the music on hold class to play while the idle
;                   channel waits for the feature to complete. If left blank,
;                   no music will be played.
;

;
; IMPORTANT NOTE: The applicationmap is not intended to be used for all Asterisk
;   applications. When applications are used in extensions.conf, they are executed
;   by the PBX core. In this case, these applications are executed outside of the
;   PBX core, so it does *not* make sense to use any application which has any
;   concept of dialplan flow. Examples of this would be things like Goto,
;   Background, WaitExten, and many more.  The exceptions to this are Gosub and
;   Macro routines which must complete for the call to continue.
;
; Enabling these features means that the PBX needs to stay in the media flow and
; media will not be re-directed if DTMF is sent in the media stream.
;
; Example Usage:
;
;testfeature => #9,peer,Playback,tt-monkeys  ;Allow both the caller and callee to play
;                                            ;tt-monkeys to the opposite channel
;
; Set arbitrary channel variables, based upon CALLERID number (Note that the application
; argument contains commas)
;retrieveinfo => #8,peer,Set(ARRAY(CDR(mark),CDR(name))=${ODBC_FOO(${CALLERID(num)})})
;
;pauseMonitor   => #1,self/callee,Pausemonitor     ;Allow the callee to pause monitoring
;                                                  ;on their channel
;unpauseMonitor => #3,self/callee,UnPauseMonitor   ;Allow the callee to unpause monitoring
;                                                  ;on their channel

; Dynamic Feature Groups:
;   Dynamic feature groups are groupings of features defined in [applicationmap]
;   that can have their own custom key mappings.  To give a channel access to a dynamic
;   feature group, add the group name to the value of the DYNAMIC_FEATURES variable.
;
; example:
; [myGroupName]         ; defines the group named myGroupName
; testfeature => #9     ; associates testfeature with the group and the keycode '#9'.
; pauseMonitor =>       ; associates pauseMonitor with the group and uses the keycode specified
;                       ; in the [applicationmap].
 
Zuletzt bearbeitet von einem Moderator:
Schau noch einmal in das Wiki, das ich Dir oben verlinkt habe, unter Configuration Options wirst Du einen eindeutigen Hinweis finden.

Beim reinkopieren Deiner sip.conf ist aber schon etwas schief gelaufen, hoffe ich, oder schaut die in Echt so aus? :shock:
 
Mache ich :) danke! Leider sieht die echt so aus :D

- - - Aktualisiert - - -

Das hat geklappt, in der featrues.cong [general] habe ich nun folgendes stehen:

pickupexten=*1
pickupsound=beep
pickupfailsound=beeperr

Das Gespräch kommt auch rüber aber irgendwie höre ich das so geholte Gespräch nicht mehr...
 
Da klappt dann wohl etwas an der RTP Aushandlung nicht, ich grabe mich aber ehrlich gesagt jetzt nicht durch die 3 Meter Config. Die solltest Du wirklich dringend ausmisten.

Der Klassiker, was so einige Audio-Probleme lösen kann, ist directmedia=no. Ansonsten müsste man in einem SIP oder RTP debug schauen, wer welche Adressen bekannt gibt.
 
gelöst - war nur ein nicht gewünschter codec ;)
 

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