[asterisk-commits] newtonr: trunk r433637 - in /trunk: ./ configs/basic-pbx/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Mar 27 16:21:08 CDT 2015


Author: newtonr
Date: Fri Mar 27 16:21:05 2015
New Revision: 433637

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=433637
Log:
configs/basic-pbx - Super Awesome Company example configs Phase 1, Patch 2

Example configuration files for a "basic PBX" deployment for the fictitious
Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4488/
and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company

Patch 4488 includes all functionality needed for SAC's outside connectivity
and some externally accessed features, as well as outbound dialing.

Reported by: Malcolm Davenport
Tested by: Rusty Newton

Review: https://reviewboard.asterisk.org/r/4488/
........

Merged revisions 433624 from http://svn.asterisk.org/svn/asterisk/branches/13

Added:
    trunk/configs/basic-pbx/cdr.conf
      - copied unchanged from r433624, branches/13/configs/basic-pbx/cdr.conf
    trunk/configs/basic-pbx/cdr_custom.conf
      - copied unchanged from r433624, branches/13/configs/basic-pbx/cdr_custom.conf
Modified:
    trunk/   (props changed)
    trunk/configs/basic-pbx/extensions.conf
    trunk/configs/basic-pbx/logger.conf
    trunk/configs/basic-pbx/modules.conf
    trunk/configs/basic-pbx/pjsip.conf

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-13-merged' - no diff available.

Modified: trunk/configs/basic-pbx/extensions.conf
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/basic-pbx/extensions.conf?view=diff&rev=433637&r1=433636&r2=433637
==============================================================================
--- trunk/configs/basic-pbx/extensions.conf (original)
+++ trunk/configs/basic-pbx/extensions.conf Fri Mar 27 16:21:05 2015
@@ -1,58 +1,169 @@
 [globals]
-; General internal dialing options used in context Internal at
-; extension dial_internal_exten. See the Dial app documentation for options.
-INTERNAL_DIAL_OPT=,10
+; General internal dialing options used in context Dial-Users.
+; Only the timeout is defined here. See the Dial app documentation for
+; additional options.
+INTERNAL_DIAL_OPT=,30
 
 [Hints]
-;Allow dynamic hint creation for every extension.
+; Allow dynamic hint creation for every extension.
 exten = _11XX,hint,PJSIP/${EXTEN}
 
 [Features]
-;Extension to check user voicemail.
+; Extension to check user voicemail. We don't requre the user to enter
+; their pincode.
 exten = 8000,1,Verbose(1, "User ${CALLERID(num)} dialed the voicemail feature.")
  same = n,VoiceMailMain(${CALLERID(num)}@example,s)
  same = n,Hangup()
 
-[DialingErrors]
-; Handle any extensions dialed that don't otherwise exist.
+; Exten to dial the main IVR internally.
+exten = 1100,1,Verbose(1, "User ${CALLERID(num)} dialed the IVR.")
+ same = n,Goto(Main-IVR,2565551100,1)
+
+[External-Features]
+; Extension for users to remotely check voicemail. Here we require the caller to
+; enter their mailbox and pincode.
+exten = 2565551234,1,Verbose(1, "User ${CALLERID(num)} dialed into remote voicemail.")
+ same = n,VoiceMailMain(example)
+ same = n,Hangup()
+
+[Dialing-Errors]
+; Handle any extensions dialed internally that don't otherwise exist.
 ; Comment out or remove this extension if you would rather have the calls
 ; ignored.
 exten = _X.,1,Verbose(1, "User ${CALLERID(num)} dialed an invalid number.")
  same = n,Playback(pbx-invalid)
  same = n,Hangup()
 
-[Internal]
-; Include other contexts providing specific functionality for internal users.
+[Internal-Setup]
+; Here we capture internal calls to do anything we need to do before sending
+; them onto all the possible internal locations. Such as disabling CDR on
+; internal to internal calls.
+exten = _X.,1,NoOp()
+ same = n,Set(CDR_PROP(disable)=1)
+ same = n,Goto(Internal-Main,${EXTEN},1)
+
+; The Internal-Main context provides a way for internal callers to get access to most
+; features and functions configured for them. External callers may be sent
+; directly to some of the contexts you see included here, so the included
+; contexts are not necessarily internal only.
+[Internal-Main]
+; The order of includes here is important for matching the right extensions.
 include = Hints
 include = Features
-include = DialingErrors
+include = Dial-Users
+include = Dialing-Errors
 
-; Handle internal calling between user extensions.
-exten = _11XX,1,Verbose(1, "User ${CALLERID(num)} dialed internally to ${EXTEN}.")
- same = n,Gosub(Internal,dial_internal_exten,1(${EXTEN}))
- same = n,Hangup()
-
-; Internal Dialing Subroutine
-;
-; ARG1 will be the extension number dialed.
-
-exten = dial_internal_exten,1,Verbose(1, "Entering internal dialing gosub.")
- same = n,Gotoif($[${DEVICE_STATE(PJSIP/${ARG1})} = BUSY]?dialed-BUSY,1:)
- same = n,Dial(PJSIP/${ARG1}${INTERNAL_DIAL_OPT})
+; Dial-Users handles calls to internal extensions.
+; Calls coming into this context may be *external* or *internal* in origin.
+[Dial-Users]
+exten = _11XX,1,Verbose(1, "User ${CALLERID(num)} dialed ${EXTEN}.")
+ same = n,Set(SAC_DIALED_EXTEN=${EXTEN})
+ same = n,Gotoif($[${DEVICE_STATE(PJSIP/${EXTEN})} = BUSY]?dialed-BUSY,1:)
+ same = n,Dial(PJSIP/${EXTEN}${INTERNAL_DIAL_OPT})
  same = n,Goto(dialed-${DIALSTATUS},1)
 
 exten = dialed-NOANSWER,1,NoOp()
- same = n,Voicemail(${ARG1}@example,u)
- same = n,Return()
+ same = n,Voicemail(${SAC_DIALED_EXTEN}@example,u)
+ same = n,Hangup()
 
 exten = dialed-BUSY,1,NoOp()
- same = n,Voicemail(${ARG1}@example,b)
- same = n,Return()
+ same = n,Voicemail(${SAC_DIALED_EXTEN}@example,b)
+ same = n,Hangup()
 
 exten = dialed-CHANUNAVAIL,1,NoOp()
  same = n,Playback(pbx-invalid)
- same = n,Return()
+ same = n,Hangup()
 
 exten = _dialed-.,1,Goto(dialed-NOANSWER,1)
 
 exten = h,1,Hangup()
+
+; Callers in the directory may dial 0 which will jump to the
+; 'o' extension.
+exten = o,1,Goto(1111)
+
+; Outbound-Dial
+;
+; Before we dial, see if the extension matches our restricted number patterns.
+; Note that this is a basic set of numbers which could incur a fee if dialed.
+; The NANP includes many other numbers that you may want to block. If you feel
+; it is necessary to block further number patterns you'll have to add them below
+; or you may consider implementing a blacklist via methods beyond the scope of
+; this example.
+[Outbound-Dial]
+exten = _011.,1,Hangup()
+exten = _900NXXXXXX,1,Hangup()
+exten = _1900NXXXXXX,1,Hangup()
+exten = _976XXXX,1,Hangup()
+exten = _NXX976XXXX,1,Hangup()
+exten = _1NXX976XXXX,1,Hangup()
+; Dial outbound through our SIP ITSP.
+exten = _X.,1,Verbose(1, "Didn't match any restricted numbers, proceeding with outbound dial.")
+ same = n,Set(CALLERID(num)=256555${CALLERID(num)})
+ same = n,Dial(PJSIP/${EXTEN}@dcs-endpoint)
+ same = n,Hangup()
+
+; Calls from internal endpoints will enter into one of the two following
+; contexts based on their dialing privilege.
+[Local]
+include = Internal-Setup
+
+exten = _NXXXXXX,1,Goto(Outbound-Dial,1256${EXTEN},1)
+exten = _256NXXXXXX,1,Goto(Outbound-Dial,1${EXTEN},1)
+exten = _1256NXXXXXX,1,Goto(Outbound-Dial,${EXTEN},1)
+
+[Long-Distance]
+include = Local
+
+exten = _NXXNXXXXXX,1,Goto(Outbound-Dial,1${EXTEN},1)
+exten = _1NXXNXXXXXX,1,Goto(Outbound-Dial,${EXTEN},1)
+
+; The DID-Extensions context captures inbound calls from our ITSP that have a
+; DID number where the last four digits matches an internal extension.
+[DID-Extensions]
+exten = _25655511XX,1,Verbose(1, "External caller dialed inbound to DID ${EXTEN})")
+ same = n,Goto(Dial-Users,${EXTEN:6},1)
+
+; Our main IVR program for receiving inbound callers.
+; The IVR script reads “Thank you for calling Super Awesome Company, Waldo’s
+; premier provider of perfect products. If you know your party’s extension,
+; you may dial it at any time. To establish a sales partnership, press one. To
+; speak with a customer advocate, press two. For accounting and other
+; receivables, press three. For a company directory, press four. For an
+; operator, press zero.”
+; demo-congrats is currently used as a placeholder.
+[Main-IVR]
+exten = 2565551100,1,Verbose(1, "New caller, ${CALLERID(num)} dialed into the IVR.")
+ same = n,Answer()
+ same = n(start),Background(demo-congrats)
+ same = n,WaitExten(10)
+ same = n,Background(demo-congrats)
+ same = n,Hangup()
+
+exten = 0,1,Verbose(1, "Caller ${CALLERID(num)} dialed the operator.")
+ same = n,Goto(Dial-Users,1111,1)
+exten = 1,1,Verbose(1, "Caller ${CALLERID(num)} dialed the Sales queue.")
+ same = n,Goto(External-Features,2565551200,1)
+exten = 2,1,Verbose(1, "Caller ${CALLERID(num)} dialed the Customer Experience queue.")
+ same = n,Goto(External-Features,2565551250,1)
+exten = 3,1,Verbose(1, "Caller ${CALLERID(num)} dialed Accounting and Receivables.")
+ same = n,Goto(Dial-Users,1106,1)
+exten = 4,1,Verbose(1, "Caller ${CALLERID(num)} dialed the directory.")
+ same = n,Directory(example,Dial-Users)
+
+exten = i,1,Playback(option-is-invalid)
+ same = n,Goto(2565551100,start)
+
+exten = t,1,Playback(are-you-still-there)
+ same = n,Goto(2565551100,start)
+
+; Calls from our ITSP SIP account arrive in DCS-Incoming. We should be careful
+; to route calls very explicitly so as to avoid any security issues, such as
+; accidentally giving outbound dial access to inbound callers.
+;
+; Each context includes extension pattern matching to match the inbound DID
+; dialed appropriately.
+[DCS-Incoming]
+include = Main-IVR
+include = DID-Extensions
+include = External-Features

Modified: trunk/configs/basic-pbx/logger.conf
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/basic-pbx/logger.conf?view=diff&rev=433637&r1=433636&r2=433637
==============================================================================
--- trunk/configs/basic-pbx/logger.conf (original)
+++ trunk/configs/basic-pbx/logger.conf Fri Mar 27 16:21:05 2015
@@ -5,5 +5,5 @@
 console = verbose,notice,warning,error
 
 ;messages = notice,warning,error
-;full = verbose,debug,notice,warning,error
+;full = verbose,notice,warning,error,debug
 ;security = security

Modified: trunk/configs/basic-pbx/modules.conf
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/basic-pbx/modules.conf?view=diff&rev=433637&r1=433636&r2=433637
==============================================================================
--- trunk/configs/basic-pbx/modules.conf (original)
+++ trunk/configs/basic-pbx/modules.conf Fri Mar 27 16:21:05 2015
@@ -1,5 +1,8 @@
 [modules]
 autoload = no
+
+; This is a minimal module load. We are loading only the modules required for
+; the Asterisk features used in the Super Awesome Company configuration.
 
 ; Applications
 
@@ -9,6 +12,7 @@
 load = app_stack.so
 load = app_verbose.so
 load = app_voicemail.so
+load = app_directory.so
 
 ; Bridging
 
@@ -18,6 +22,10 @@
 load = bridge_native_rtp.so
 load = bridge_simple.so
 load = bridge_softmix.so
+
+; Call Detail Records
+
+load = cdr_custom.so
 
 ; Channel Drivers
 
@@ -41,9 +49,11 @@
 ; Functions
 
 load = func_callerid.so
+load = func_cdr.so
 load = func_pjsip_endpoint.so
 load = func_sorcery.so
 load = func_devstate.so
+load = func_strings.so
 
 ; Core/PBX
 

Modified: trunk/configs/basic-pbx/pjsip.conf
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/basic-pbx/pjsip.conf?view=diff&rev=433637&r1=433636&r2=433637
==============================================================================
--- trunk/configs/basic-pbx/pjsip.conf (original)
+++ trunk/configs/basic-pbx/pjsip.conf Fri Mar 27 16:21:05 2015
@@ -9,13 +9,58 @@
 ;external_media_address = 203.0.113.1
 ;external_signaling_address = 203.0.113.1
 
+;================================ CONFIG FOR SIP ITSP ==
+
+; Registration for Digium Cloud Services Account
+
+[dcs-trunk]
+type = registration
+transport = transport-udp-nat
+outbound_auth = dcs-trunk-auth
+server_uri = sip:sip.digiumcloud.net
+;client_uri = sip:myaccountID at sip.digiumcloud.net
+retry_interval = 60
+
+[dcs-trunk-auth]
+type = auth
+auth_type = userpass
+;username = myaccountID
+;password = ASTRONGPASSWORD
+
+; Endpoint for Digium Cloud Services account
+
+[dcs-endpoint]
+type=endpoint
+transport = transport-udp-nat
+context = DCS-Incoming
+allow = !all,g722,ulaw
+outbound_auth = dcs-auth
+aors = dcs-aor
+direct_media = no
+from_domain = sip.digiumcloud.net
+
+[dcs-auth]
+type = auth
+auth_type = userpass
+;username = myaccountID
+;password = ASTRONGPASSWORD
+realm = sip.digiumcloud.net
+
+[dcs-aor]
+type = aor
+contact = sip:sip.digiumcloud.net
+
+[dcs-identify]
+type=identify
+endpoint = dcs-endpoint
+;match = 8.17.32.12
 
 ;================================ ENDPOINT TEMPLATES ==
 ; Our primary endpoint template for internal desk phones.
 [endpoint-internal-d70](!)
 type = endpoint
 transport = transport-udp-nat
-context = Internal
+context = Long-Distance
 allow = !all,g722,ulaw
 direct_media = no
 trust_id_outbound = yes
@@ -110,7 +155,7 @@
 [1103](endpoint-internal-d70)
 auth = 1103
 aors = 1103
-callerid = Penelope Bronte <103>
+callerid = Penelope Bronte <1103>
 
 [1103](auth-userpass)
 password = zutAnacHe8ewuWr




More information about the asterisk-commits mailing list