#9621: Add asterisk configuration files for functional tests on Jenkins

tools/asterisk/extensions.conf

+; 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				; Console interface for demo
+;CONSOLE=DAHDI/1
+;CONSOLE=Phone/phone0
+IAXINFO=guest					; IAXtel username/password
+;IAXINFO=myuser:mypass
+TRUNK=DAHDI/G2					; Trunk interface
+;
+; Note the 'G2' in the TRUNK variable above. It specifies which group (defined
+; in chan_dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use
+; in the specified group. The four possible options are:
+;
+; g: select the lowest-numbered non-busy DAHDI channel
+;    (aka. ascending sequential hunt group).
+; G: select the highest-numbered non-busy DAHDI channel
+;    (aka. descending sequential hunt group).
+; r: use a round-robin search, starting at the next highest channel than last
+;    time (aka. ascending rotary hunt group).
+; R: use a round-robin search, starting at the next lowest channel than last
+;    time (aka. descending rotary hunt group).
+;
+TRUNKMSD=1					; MSD digits to strip (usually 1 or 0)
+;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.
+;
+; 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(${ARG1})
+exten => s,n(fail),Hangup
+
+
+[demo]
+include => stdexten
+;
+; We start with what to do when a call first comes in.
+;
+exten => s,1,Wait(1)			; Wait a second, just for fun
+exten => s,n,Answer			; Answer the line
+exten => s,n,Set(TIMEOUT(digit)=5)	; Set Digit Timeout to 5 seconds
+exten => s,n,Set(TIMEOUT(response)=10)	; Set Response Timeout to 10 seconds
+exten => s,n(restart),BackGround(demo-congrats)	; Play a congratulatory message
+exten => s,n(instruct),BackGround(demo-instruct)	; Play some instructions
+exten => s,n,WaitExten			; Wait for an extension to be dialed.
+
+exten => 2,1,BackGround(demo-moreinfo)	; Give some more information.
+exten => 2,n,Goto(s,instruct)
+
+exten => 3,1,Set(LANGUAGE()=fr)		; Set language to french
+exten => 3,n,Goto(s,restart)		; Start with the congratulations
+
+exten => 1000,1,Goto(default,s,1)
+;
+; We also create an example user, 1234, who is on the console and has
+; voicemail, etc.
+;
+exten => 1234,1,Playback(transfer,skip)		; "Please hold while..."
+					; (but skip if channel is not up)
+exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
+exten => 1234,n,Goto(default,s,1)		; exited Voicemail
+
+exten => 1235,1,Voicemail(1234,u)		; Right to voicemail
+
+exten => 1236,1,Dial(Console/dsp)		; Ring forever
+exten => 1236,n,Voicemail(1234,b)		; Unless busy
+
+;
+; # for when they're done with the demo
+;
+exten => #,1,Playback(demo-thanks)	; "Thanks for trying the demo"
+exten => #,n,Hangup			; Hang them up.
+
+;
+; A timeout and "invalid extension rule"
+;
+exten => t,1,Goto(#,1)			; If they take too long, give up
+exten => i,1,Playback(invalid)		; "That's not valid, try again"
+
+;
+; Create an extension, 500, for dialing the
+; Asterisk demo.
+;
+exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
+exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default)	; Call the Asterisk demo
+exten => 500,n,Playback(demo-nogo)	; Couldn't connect to the demo site
+exten => 500,n,Goto(s,6)		; Return to the start over message.
+
+;
+; Create an extension, 600, for evaluating echo latency.
+;
+exten => 600,1,Playback(demo-echotest)	; Let them know what's going on
+exten => 600,n,Echo			; Do the echo test
+exten => 600,n,Playback(demo-echodone)	; Let them know it's over
+exten => 600,n,Goto(s,6)		; Start over
+
+;
+;	You can use the Macro Page to intercom a individual user
+exten => 76245,1,Macro(page,SIP/Grandstream1)
+; or if your peernames are the same as extensions
+exten => _7XXX,1,Macro(page,SIP/${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/${EXTEN})
+
+;[mainmenu]
+;
+; Example "main menu" context with submenu
+;
+;exten => s,1,Answer
+;exten => s,n,Background(thanks)		; "Thanks for calling press 1 for sales, 2 for support, ..."
+;exten => s,n,WaitExten
+;exten => 1,1,Goto(submenu,s,1)
+;exten => 2,1,Hangup
+;include => default
+;
+;[submenu]
+;exten => s,1,Ringing					; Make them comfortable with 2 seconds of ringback
+;exten => s,n,Wait,2
+;exten => s,n,Background(submenuopts)	; "Thanks for calling the sales department.  Press 1 for steve, 2 for..."
+;exten => s,n,WaitExten
+;exten => 1,1,Goto(default,steve,1)
+;exten => 2,1,Goto(default,mark,2)
+
+[default]
+;
+; By default we include the demo.  In a production system, you
+; probably don't want to have the demo there.
+;
+include => demo
+
+exten => 100,1,Dial(SIP/100)
+exten => 200,1,Dial(SIP/200)
+exten => 300,1,Dial(SIP/300)
+
+;
+; An extension like the one below can be used for FWD, Nikotel, sipgate etc.
+; Note that you must have a [sipprovider] section in sip.conf
+;
+;exten => _41X.,1,Dial(SIP/${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/1@192.168.0.14)
+;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
+;exten => 6391,1,Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org) ;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(${WIL}))
+						; Ditto for wil
+;exten => 6536,n,Goto(default,s,1)		; exited Voicemail
+;exten => wil,1,Goto(6236,1)
+
+;If you want to subscribe to the status of a parking space, this is
+;how you do it. Subscribe to extension 6600 in sip, and you will see
+;the status of the first parking lot with this extensions' help
+;exten => 6600,hint,park:701@parkedcalls
+;exten => 6600,1,noop
+;
+; Some other handy things are an extension for checking voicemail via
+; voicemailmain
+;
+;exten => 8500,1,VoicemailMain
+;exten => 8500,n,Hangup
+;
+; Or a conference room (you'll need to edit meetme.conf to enable this room)
+;
+;exten => 8600,1,Meetme(1234)
+;
+; Or playing an announcement to the called party, as soon it answers
+;
+;exten = 8700,1,Dial(${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.

tools/asterisk/sip.conf

+;
+; 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 - IP address filters
+;		- contactpermit/contactdeny - 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::account02@domain.com:5062
+;        SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
+;
+; 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@mysipproxy!sales@edvina.net
+;
+; 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]
+context=default                 ; Default context for incoming calls
+;allowguest=no                  ; Allow or reject guest calls (default is yes)
+				; If your Asterisk is connected to the Internet
+				; and you have allowguest=yes
+				; you want to check which services you offer everyone
+				; out there, by enabling them in the default context (see below).
+;match_auth_username=yes        ; if available, match user entry using the
+                                ; 'username' field from the authentication line
+                                ; instead of the From: field.
+allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
+;allowtransfer=no               ; Disable all transfers (unless enabled in peers or users)
+                                ; Default is enabled. The Dial() options 't' and 'T' are not
+                                ; related as to whether SIP transfers are allowed or not.
+;realm=mydomain.tld             ; Realm for digest authentication
+                                ; defaults to "asterisk". If you set a system name in
+                                ; asterisk.conf, it defaults to that system name
+                                ; Realms MUST be globally unique according to RFC 3261
+                                ; Set this to your host name or domain name
+;domainsasrealm=no              ; Use domans list as realms
+                                ; You can serve multiple Realms specifying several
+                                ; 'domain=...' directives (see below).
+                                ; In this case Realm will be based on request 'From'/'To' header
+                                ; and should match one of domain names.
+                                ; Otherwise default 'realm=...' will be used.
+
+; With the current situation, you can do one of four things:
+;  a) Listen on a specific IPv4 address.      Example: bindaddr=192.0.2.1
+;  b) Listen on a specific IPv6 address.      Example: bindaddr=2001:db8::1
+;  c) Listen on the IPv4 wildcard.            Example: bindaddr=0.0.0.0
+;  d) Listen on the IPv4 and IPv6 wildcards.  Example: bindaddr=::
+; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
+; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
+; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
+;  IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
+;
+; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
+; for TLS).
+;   IPv4 example: bindaddr=0.0.0.0:5062
+;   IPv6 example: bindaddr=[::]:5062
+;
+; The address family of the bound UDP address is used to determine how Asterisk performs
+; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only
+; AAAA records are considered. In case d), both A and AAAA records are considered. Note,
+; however, that Asterisk ignores all records except the first one. In case d), when both A
+; and AAAA records are available, either an A or AAAA record will be first, and which one
+; depends on the operating system. On systems using glibc, AAAA records are given
+; priority.
+
+udpbindaddr=0.0.0.0:5062             ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
+                                     ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
+
+; When a dialog is started with another SIP endpoint, the other endpoint
+; should include an Allow header telling us what SIP methods the endpoint
+; implements. However, some endpoints either do not include an Allow header
+; or lie about what methods they implement. In the former case, Asterisk
+; makes the assumption that the endpoint supports all known SIP methods.
+; If you know that your SIP endpoint does not provide support for a specific
+; method, then you may provide a comma-separated list of methods that your
+; endpoint does not implement in the disallowed_methods option. Note that
+; if your endpoint is truthful with its Allow header, then there is no need
+; to set this option. This option may be set in the general section or may
+; be set per endpoint. If this option is set both in the general section and
+; in a peer section, then the peer setting completely overrides the general
+; setting (i.e. the result is *not* the union of the two options).
+;
+; Note also that while Asterisk currently will parse an Allow header to learn
+; what methods an endpoint supports, the only actual use for this currently
+; is for determining if Asterisk may send connected line UPDATE requests. Its
+; use may be expanded in the future.
+;
+; disallowed_methods = UPDATE
+
+;
+; Note that the TCP and TLS support for chan_sip is currently considered
+; experimental.  Since it is new, all of the related configuration options are
+; subject to change in any release.  If they are changed, the changes will
+; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
+;
+tcpenable=no                    ; Enable server for incoming TCP connections (default is no)
+tcpbindaddr=0.0.0.0             ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
+                                ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
+
+;tlsenable=no                   ; Enable server for incoming TLS (secure) connections (default is no)
+;tlsbindaddr=0.0.0.0            ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
+                                ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
+                                ; Remember that the IP address must match the common name (hostname) in the
+                                ; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
+                                ; For details how to construct a certificate for SIP see
+                                ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
+
+;tcpauthtimeout = 30            ; tcpauthtimeout specifies the maximum number
+				; of seconds a client has to authenticate.  If
+				; the client does not authenticate beofre this
+				; timeout expires, the client will be
+                                ; disconnected. (default: 30 seconds)
+
+;tcpauthlimit = 100             ; tcpauthlimit specifies the maximum number of
+				; unauthenticated sessions that will be allowed
+                                ; to connect at any given time. (default: 100)
+
+srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
+                                ; 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
+                                ; and subscriptions (seconds)
+;minexpiry=60                   ; Minimum length of registrations/subscriptions (default 60)
+;defaultexpiry=120              ; Default length of incoming/outgoing registration
+;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
+;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
+;
+; 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
+;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.
+;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]
+
+;maxcallbitrate=384             ; Maximum bitrate for video calls (default 384 kb/s)
+                                ; Videosupport and maxcallbitrate is settable
+                                ; for peers and users as well
+;callevents=no                  ; generate manager events when sip ua
+                                ; performs events (e.g. hold)
+;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.
+
+;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
+;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.
+
+;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
+
+; 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.
+;
+;------------------------ TLS settings ------------------------------------------------------------
+;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem format only) to use for TLS connections
+                                        ; 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'.
+;
+;session-timers=originate
+;session-expires=600
+;session-minse=90
+;session-refresher=uas
+;
+;--------------------------- SIP DEBUGGING ---------------------------------------------------
+;sipdebug = yes                 ; Turn on SIP debugging by default, from
+                                ; the moment the channel loads this configuration
+;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:password@mysipprovider.com
+;
+;     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:xxxxxx@sip-tls-proxy.example.org
+;
+;    The 'transport' part defaults to 'udp' but may also be 'tcp' or 'tls'.
+;    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
+
+;----------------------------------------- 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:password@mysipprovider.com/1234
+;mwi => 1234:password@myportprovider.com:6969/1234
+;mwi => 1234:password:authuser@myauthprovider.com/1234
+;mwi => 1234:password:authuser@myauthportprovider.com: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                ; Default. Use rport if the remote side says to use it.
+;        nat = force_rport       ; Force rport to always be on.
+;        nat = yes               ; Force rport to always be on and perform comedia RTP handling.
+;        nat = comedia           ; Use rport if the remote side says to use it and perform comedia RTP handling.
+;
+; '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 port number they tell Asterisk to send RTP packets to
+; for their media streams is not actual port number that will be used on the nearer
+; side of the NAT.
+;
+; 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
+
+;----------------------------------- 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'.
+
+;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.
+
+;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.
+
+;----------------------------------------- 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:topsecret@digium.com
+;
+; You may also add auth= statements to [peer] definitions
+; Peer auth= override all other authentication settings if we match on realm
+
+;------------------------------------------------------------------------------
+; DEVICE CONFIGURATION
+;
+; The SIP channel has two types of devices, the friend and the peer.
+; * The type=friend is a device type that accepts both incoming and outbound calls,
+;   where Asterisk match on the From: username on incoming calls.
+;   (A synonym for friend is "user"). This is a type you use for your local
+;   SIP phones.
+; * The type=peer also handles both incoming and outbound calls. On inbound calls,
+;   Asterisk only matches on IP/port, not on names. This is mostly used for SIP
+;   trunks.
+;
+; 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
+; insecure
+; trustrpid
+; progressinband
+; promiscredir
+; useclientcode
+; accountcode
+; setvar
+; callerid
+; amaflags
+; callcounter
+; busylevel
+; allowoverlap
+; allowsubscribe
+; allowtransfer
+; ignoresdpversion
+; subscribecontext
+; template
+; videosupport
+; maxcallbitrate
+; rfc2833compensate
+; mailbox
+; session-timers
+; session-expires
+; session-minse
+; session-refresher
+; t38pt_usertpsource
+; regexten
+; fromdomain
+; fromuser
+; host
+; port
+; qualify
+; defaultip
+; defaultuser
+; rtptimeout
+; rtpholdtimeout
+; sendrpid
+; outboundproxy
+; rfc2833compensate
+; callbackextension
+; registertrying
+; 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,
+;                       ; then call oneself, and get redirected to that
+;                       ; same location).
+; directmediapermit
+; directmediadeny
+; unsolicited_mailbox
+; use_q850_reason
+; maxforwards
+; encryption
+
+;[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
+        nat=yes
+        directmedia=no
+        host=dynamic
+
+[public-phone](!,basic-options)   ; another template inheriting basic-options
+        nat=no
+        directmedia=yes
+
+[my-codecs](!)                    ; a template for my preferred codecs
+        disallow=all
+        allow=ilbc
+        allow=g729
+        allow=gsm
+        allow=g723
+        allow=ulaw
+
+[ulaw-phone](!)                   ; and another one for ulaw-only
+        disallow=all
+        allow=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
+;callerid=John Doe <1234>        ; Full caller ID, to override the phones config
+                                 ; on incoming calls to Asterisk
+;host=192.168.0.23               ; we have a static but private IP address
+                                 ; No registration allowed
+;nat=no                          ; there is not NAT between phone and Asterisk
+;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 README.callingpres for more information
+
+;[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
+;nat=yes                         ; X-Lite is behind a NAT router
+;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       ; 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
+;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.
+
+;[cisco1]
+;type=friend
+;secret=blah
+;qualify=200                     ; Qualify peer is no more than 200ms away
+;nat=yes                         ; This phone may be natted
+                                 ; Send SIP and RTP to the IP address that packet is
+                                 ; received from instead of trusting SIP headers
+;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.
+
+;[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.
+
+[100]
+type=friend
+host=dynamic
+username=100
+secret=password
+canreinvite=no
+allow=all
+
+[200]
+type=friend
+host=dynamic
+username=200
+secret=password
+canreinvite=no
+allow=all
+
+[300]
+type=friend
+host=dynamic
+username=300
+canreinvite=no
+allow=all