SIP Headers – Compact Form
SIP Headers – Compact Form Many common SIP header fields have a compact form where the header field name is
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sip
Basic Call Flow – SIP
eduguru asterisk, Basic Call Flow - SIP, Example to understand the term, Explanation of above peer to peer sip communication, Key terms used in SIP call flow, sip, sip communication, sip protocall
Basic Call Flow – SIP Key terms used in SIP: It is important to familiarize with these terms in order
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Asterisk Dial () command
Satya Prakash 0 Comments '15', [URL]]]) Arguments Technology/Resource Technology/Resource - Specification of the device(s) to dial. These must be in the format of Technology/Resource, 1, 10) When the two channels are connected together ("bridged") allowing a conversation to take place between them. When the channel that triggered the Dial command hangs up, 15) exten => s, 20, 20) exten => _908., 3, a call to a SIP device (as defined in sip.conf) might have a destination of SIP/Jane, a forward slash, always set HANGUPCAUSE to 'answered elsewhere' d - Allow the calling user to dial a 1 digit extension while waiting for a call to be answered. Exit to that extension if it exists in the current contex, and a call to an IAX device (defined in iax.conf) might have a destination of IAX2/Fred. If we wanted Asterisk to ring the Zap/1 channel when extension 123 is reached in the dialplan, and bridge the inbound call with whichever destination channel answers the call first. If the Dial () application can’t contact any of the destinations, and continue on with the next priority in the extension. This application sets the following channel variables: DIALEDTIME - This is the time from dialing a channel until when it is disconnected. ANSW, and IAX2. For example, and Resource represents a resource available to that particular channel driver. Technology2/Resource2 - Optional extra devices to dial in parallel If you need more then one enter them as Technology2/R, and the calling DTMF string is sent to the calling party. Both arguments can be used alone. If progress is specified, and the remote endpoint or resource. Common technology types include Zap (for analog and T1/E1/J1 channels), and the resource is 1. Similarly, as shown: Dial(Zap/[gGrR]channel_or_group[/remote_extension]) For example, asterisk dial, Asterisk Dial () command, asterisk dial option, Asterisk takes most of the hard work out of connecting and translating between disparate networks. All you have to do is learn how to use the Dial() application. The syntax of the Dial() application i, Asterisk will set a variable called DIALSTATUS and then continue on with the next priority in the extension. Example: exten => 1265, Asterisk will set a variable called DIALSTATUS with the reason that it couldn’t dial the destinations, asterisl dial with option, but before the call gets bridged. The called DTMF string is sent to the called party, but the switch will not release their line until the destination party (the operator) hangs up. mode - With mode either not specified or set to 1, but when options such as A() and M() are used, by concatenating the destinations with an ampersand (&), caller A might be communicating over the traditional analog telephone network, cancel any dial timeout which has been set for this call. URL - The optional URL will be sent to the called party if the channel driver supports it. As an example, crh) exten => 20, Dial() will attempt to call the destination(s) for that number of seconds before giving up and moving on to the next priority in the extension. If no timeout is specified, Dial() will continue to dial the called channel(s) until someone answers or the caller hangs up. Let’s add a timeout of 10 seconds to our extension: exten => 123, Dial(IAX2/guest@misery.digium.com/s) The full syntax for the Dial() application is slightly different when dealing with Zap channels, Dial(Modem/ttyI0:${EXTEN:1}) exten => 233, Dial(Phone/phone0, Dial(SIP/1234, Dial(SIP/4029&SIP/4027&Zap/4&IAX/jaz, Dial(Zap/1, Dial(Zap/1)\ We can also dial multiple channels at the same time, Dial(Zap/1&Zap/2&SIP/Jane) The Dial() application will ring the specified destinations simultaneously, Dial(Zap/2r2, Dial(Zap/3/5551234), Dial(Zap/4/18005551212) The second argument to the Dial() application is a timeout, do not screen the call. o - If x is not provided, even if the called party isn't actually ringing. Pass no audio to the calling party until the called channel has answered. tone - Indicate progress to calling party. Send audio 'tone' from indications, filename specifies the sound prompt to play as a warning when time x is reached. If not set, filename specifies the sound prompt to play when the call begins. If not set, filename specifies the sound prompt to play when the timeout is reached. If not set, force the CallerID sent on a call-forward or deflection to the dialplan extension of this Dial() using a dialplan hint. For example, force the CallerID sent to x. x F - When the caller hangs up, Gosub to the specified location using the current channel. context exten priority arg1 argN C - Reset the call detail record (CDR) for this call. c - If the Dial() application cancels this call, Gosub to the specified location using the newly created channel. The Gosub will be executed for each destination channel. context exten priority arg1 argN B - Before initiating the outgoing call(s), here is how you would dial 1-800-555-1212 on Zap channel number 4. exten => 501, if it exists. D - Send the specified DTMF strings after the called party has answered, in milliseconds m - Provide hold music to the calling party until a requested channel answers. A specific music on hold class (as defined in musiconhold.conf) can be specified. class M - Execute the s, in milliseconds y - Warning time, in milliseconds z - Repeat time, is busy, it will be ignored. When the destination answers (presumably an operator services station), it will ring their phone back. p - This option enables screening mode. This is basically Privacy mode without memory. P - Enable privacy mode. Use x as the family/key in the AstDB database if it is pr, its DTMF is sent immediately after receiving a PROGRESS message. called calling progress e - Execute the h extension for peer after the call ends f - If x is not provided, let’s assume that we want to call a Zap endpoint identified by Zap/1, like this: exten => 123, options, or is otherwise unavailable, or the context defined in the EXITCONTEXT variable, sip, some PSTNs do not allow CallerID to be set to anything other than the numbers assigned to you. If x is provided, specified in seconds. If a timeout is given, specify that the CallerID that was present on the calling channel be stored as the CallerID on the called channel. This was the behavior of Asterisk 1.0 and earlier. If x is provided, specify the CallerID stored on the called channel. Note that o(${CALLERID(all)}) is similar to option o without the parameter. x O - Enables operator services mode. This option only works when bridgin, the calling channel is answered when the called channel answers, the calling channel is not answered until all actions on the called channel (such as playing an announcement) are completed. This option can be used to answer the calling channel before doing anything, the channels are bridged and the dialplan is done. If the destination simply does not answer, the default behavior is adequate in most cases. b - Before initiating an outgoing call, the Dial command exits. If the call is answered before the timeout, the introduction will always be deleted. N - This option is a modifier for the call screening/privacy mode. It specifies that if Caller*ID is present, the originator hanging up will cause the phone to ring back immediately. With mode set to 2, the originator no longer has control of their line. They may hang up, the recorded introduction will not be deleted if the caller hangs up while the remote party has not yet answered. With delete set to 1, the time remaining will be announced. FILENAME LIMIT_CONNECT_FILE - If specified, the time remaining will be announced. FILENAME LIMIT_WARNING_FILE - If specified, the time remaining will be announced. FILENAME x - Maximum call time, this action will be taken after the macro finished executing. ABORT - Hangup both legs of the call CONGESTION - Behave as if line congestion was encountered BUSY - Behave as if a busy signal was encou, this defaults to 136 years. options A - Play an announcement to the called party, this variable causes Asterisk to play the prompts to the callee. YES NO default: (true) LIMIT_TIMEOUT_FILE - If specified, this variable causes Asterisk to play the prompts to the caller. YES default: (true) NO LIMIT_PLAYAUDIO_CALLEE - If set, timeout, transfer the called party to the next priority of the current extension and start execution at that location. g - Proceed with dialplan execution at the next priority in the current extension if the d, transfer the called party to the specified destination and start execution at that location. context exten priority F - When the caller hangs up, transfer the calling party to the specified priority and the called party to the specified priority plus one. context exten priority h - Allow the called party to hang up by sending the DTMF sequence , tTr) exten => 500, we’d add the following extension: exten => 123, What is command and how it works? One of Asterisk’s most valuable features is its ability to connect different callers to each other. This is especially useful when callers are using different methods, when the operator flashes the trunk, where Technologyrepresents a particular channel driver, where x is the prompt to be played x - The file to play to the called party a - Immediately answer the calling channel when the called channel answers in all cases. Normally, which (in its simplest form) is made up of a technology (or transport) across which to make the call, which is an FXS channel with an analog phone plugged into it. The technology is Zap, while user B might be sitting in a café halfway around the world and speaking on an IP telephone. Luckily, you can dial into a demonstration server at Digium using the IAX2 protocol by using the following extension: exten => 500
What is command and how it works? One of Asterisk’s most valuable features is its ability to connect different callers
Read moreAsterisk DialStatus Channel Varriable
eduguru 1 Comment ${ARG3}, ${ARG4}) exten => s, 1, 1) exten => s-CANCEL, 1) exten => s-NOANSWER, 2, arg2, Asterisk channel variable DIALSTATUS Contains a text string signifying result of the last dial attempt: ANSWER: Call is answered. A successful dial. The caller reached the callee. BUSY: Busy signal. T, Asterisk DialStatus Channel Varriable, Busy exten => s-CHANUNAVAIL, callee chose to send caller to torture menu INVALIDARGS: Error parsing Dial command arguments (added for Asterisk 1.4.1, callee rejected the call TORTURE: Privacy mode, Congestion exten => _s-., Congestion exten => s-, Congestion exten => s-CANCEL, Congestion The following is an example of a local Zap channel callout with SIP failover (ex. Zap channel is in use) [macro-localcallout] exten => s, Dial() returns DIALSTATUS=CONGESTION for pretty much every call setup problem. The reason for this is that Dial() is used for multiple protocols (Zap, Dial(${ARG1}, Dial(${ZAP/1/${ARG1}, Dial(SIP/sippeer/${LOCALAREACODE}${ARG1}, g, Goto(s-${DIALSTATUS}, GotoIf($["${DIALEDTIME}" = "0"]?here) exten => s-NOANSWER, GotoIf($["${DTIME}" = "0"]?here) exten => s-NOANSWER, Hangup exten => s-BUSY, Hangup exten => s-CONGESTION, Hangup exten => s-NOANSWER, IAX etc.) and so it is limited to the lowest common denominator and is unable to return the protocol specific information (e.g., in sip.conf or iax.conf). CONGESTION "Congestion" is somewhat misleading. Unfortunately at this time (2007-05-17), n, n(here), Need failover for "${ARG1}") exten => s-BUSY, Need failover for "${ARG1}") exten => s-CONGESTION, next line is not perfect! ;exten => s-NOANSWER, NoOp( Dial Status: ${DIALSTATUS}) exten => s, peer may not be registered. DONTCALL: Privacy mode, see example in goto documentation. Note: To obtain useful DIALSTATUS information when dialing a peer the peer's definition must contain qualify=yes (e.g., Set(DIALSTART=${EPOCH}) ;exten => s, Set(DTIME=$[${EPOCH} - ${DIALSTART}]) watch out DIALEDTIME can also be empty, SetCallerId(${CALLERIDNUM}) exten => s-CHANUNAVAIL, sip, SIP 404 response) ., SVN r53135-53136) The dial status may be used in the dialplan to control program flow, T), T) exten => s, the number rang for too long, then the dial timed out. CANCEL: Call is cancelled. The dial command reached its number but the caller hung up before the callee picked up. CONGESTION: Congestion. This status is usually a sign that t, This macro is an example of how failover for two or more outgoing trunks can be managed [macro-safedial] exten => s, Verbose(1
Asterisk channel variable DIALSTATUS Contains a text string signifying result of the last dial attempt: ANSWER: Call is answered. A
Read moreastrerisk dialplan configuration file – extensions.conf
eduguru 1 Comment , , ; extensions.conf - the Asterisk dial plan ; ; Static extension configuration file, ; Phone, ; while _1NXXNXXXXXX would represent an area code plus phone number ; preceded by a on. ; ; Each step of an extension is ordered by priority, ; you have to goto "s" to execute that extension. ; ; Note: In old versions of Asterisk the PBX in some cases defaulted to ; extension "s" when a given extension was wrong (like in AMI originate). ; T, ; your global variables may be overridden by that file. Please take care to ; use only one location to set global variables, ..." ;exten => s, ...) ; ; Timing list for includes is ; ;, [, ] ; ; Note that ranges may be specified to wrap around the ends. Also, {priority|label{+|-}offset}[(alias)], *, * ; ; 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 , * ;include => weekend, * ;include => weeknights, ${EXTEN:${GLOBAL(TRUNKMSD)}})}) [trunkld] ; ; Long distance context accessed through trunk ; exten => _91NXXNXXXXXX, ${EXTEN:1}) exten => _91NXXNXXXXXX, ${EXTEN:2})}@sipprovider, ${EXTEN:4}) exten => _9011., ${EXTEN}, ${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION, ${timezone}, 000 extensions, 1, 1 extension-priority) ; ${ARG4} - Optional TORTURE context name to jump to (assumes the s, 1 extension-priority)` ; ${ARG5} - Context in voicemail (if empty, 1) ; ; We also create an example user, 1) ; alias mark to 6275 ;exten => 6536, 1) ; Callee chose to send this call to a polite "Don't call again" script. exten => stdexten-TORTURE, 1) ; Callee chose to send this call to a telemarketer torture script. exten => _stde[x]te[n]-., 1) ; exited Voicemail ;exten => _125642860XX, 1) ; exited Voicemail ;exten => mark, 1) ; exited Voicemail ;exten => wil, 1) ; exited Voicemail exten => 1235, 1) ; filter out bad characters per the README-SERIOUSLY.best-practices.txt document same => n, 1) ; If they take too long, 1) ; Jump based on status (NOANSWER, 1) ; Treat anything else as no answer exten => a, 1) ;exten => 2, 1) ;If you want to subscribe to the status of a parking space, 1) [outbound-freenum2] ; This is the handler which performs the dialing logic. It is called ; from the [outbound-freenum] context ; exten => _X!, 1) exten => _XX*X!, 1) exten => _XXX*X!, 1) exten => _XXXX*X!, 1) exten => _XXXXX*X!, 1) exten => _XXXXXX*X!, 1) exten => fn-BUSY, 1) exten => s-NOANSWER, 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, 1) same => n(from), 1234, 15) ;exten => 1265, 17:02-8:58, 2, 2 ;exten => s, 2 for support, 2 for..." ;exten => s, 2-)}) ; make sure the suffix is all digits as well same => n, 2) [public] ; ; ATTENTION: If your Asterisk is connected to the internet and you do ; not have allowguest=no in sip.conf, 20, 20 seconds maximum, 20 seconds maximum exten => _X., 20) ; Ring the interface, 3, 30, 30000(time), 40) same => n, 40000(ani), 5, 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, 500, 50000(stdexten), 6, 6) ; ; Here's what a phone entry would look like (IXJ for example) ; ;exten => 1265, 6) ; Return to the start over message. ; ; Create an extension, 6) ; Start over ; ; You can use the Macro Page to intercom a individual user exten => 76245, 600, 60000(stdPrivacyexten), 7, 8, 9:00-17:00, 9) ; . - wildcard, a "reload" will often leave the globals vars cleared, 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, acme-incoming) ; ; grouping of acme's extensions... never used directly, acme)) ;exten => 111, acme)) ;exten => 112, 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 ', always included. ; ;[acme-extens] ;include => stdexten ;exten => 111, and ; switches. Includes are always processed depth-first. So for example, and ; the desperate. If you have large dialplans (more than about 50 extensions ; in a context), and are forced to return to the ; old pattern matcher, and even if deleted from the extensions.conf or ; one of its included files, 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: sele, and iaxtel calls only ; ignorepat => 9 include => default include => trunklocal include => iaxtel700 include => trunktollfree include => iaxprovider ;Include parkedcalls (or the cont, and its value will revert to its ; previous value (before being declared as LOCAL()) upon Return. ; exten => _X., and other places where ; the timezone is significant (e.g. calls to "SayUnixTime()", and Others exten => s, 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 throug, and then not set, and they ; will be inserted where you insert the #exec command. The #exec command ; works on all asterisk configuration files. However, and to sign ; up, and use global variables ;exten => 6245, and VPB. Other channels, and you will likely save ; yourself a ton of grief. ; WARNING WARNING WARNING WARNING ; ; Any category other than "General" and "Globals" represent ; extension contexts, and you will see ;the status of the first parking lot with this extensions' help ;exten => 6600, and/or high call volume, Answer ; Answer the line exten => s, Answer ;exten => s, 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., Answer() exten => _X., Answer() exten => 4339840, ANSWER) exten => stdexten-NOANSWER, application(arg1, 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, arg2, as soon it answers ; ;exten = 8700, 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, astrerisk dialplan configuration file - extensions.conf, b) ; If busy, b) ; Unless busy ; ; # for when they're done with the demo ; exten => #, b) ; Voicemail (busy) ;exten => 6361, BackGround(demo-congrats) ; Play a congratulatory message exten => s, BackGround(demo-instruct) ; Play some instructions exten => s, BackGround(demo-moreinfo) ; Give some more information. exten => 2, Background(submenuopts) ; "Thanks for calling the sales department. Press 1 for steve, Background(thanks) ; "Thanks for calling press 1 for sales, Background(vm-enter-num-to-call) ;exten => s, 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=defau, 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, BUSY, Busy() exten => _f[n]-., but ; there are exceptions. For more information, call screening ; option (or use P for databased call _X.creening) exten => _X., causes the matching process to complete as soon as ; it can unambiguously determine that no other matches are possible ; ; For example, ChanIsAvail(${ARG1}, CHANUNAVAIL, CONGESTION, CONGESTION ; or HANGUP depending on Asterisk's best guess. This is the default. ; ; If autofallthrough is not set, Congestion() [macro-trunkdial] ; ; Standard trunk dial macro (hangs up on a dialstatus that should ; terminate call) ; ${ARG1} - What to dial ; exten => s, d) ; Give voicemail at extension 8500 ; exten => 8500, Dial(${ARG1}) exten => s, Dial(${dev}, Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [international] ; ; Master context for international long distance ; ignorepat => 9 include => longdistance include => trunkint [longdistan, Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [trunklocal] ; ; Local seven-digit dialing accessed through trunk interface ; exten => _9NXXXXXX, Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [trunktollfree] ; ; Long distance context accessed through trunk interface ; exten => _91800NXXXXXX, Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) exten => _91866NXXXXXX, Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) exten => _91877NXXXXXX, Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) exten => _91888NXXXXXX, Dial(${GLOBAL(TRUNK)}/${FILTER(0-9, Dial(${HINT}, Dial(${MARK}, Dial(${TRUNK}/${EXTEN}, Dial(Console/dsp) ; Ring forever exten => 1236, Dial(DAHDI/g0/${EXTEN}, Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325 [dundi-e164-local] ; ; Context to put your dundi IAX2 or SIP user in for ; full access ; include => dundi-e164-canonical include => dundi-e164-custo, Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428 ;exten => _1256325XXXX, 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 ; , Dial(IAX2/customer2) [dundi-e164-via-pstn] ; ; If you are freely delivering calls to the PSTN, Dial(IAX2/guest@pbx.digium.com/s@default) ; Call the Asterisk demo exten => 500, Dial(IAX2/JaneDoe, Dial(IAX2/otherbox/${EXTEN:7}) [dundi-e164-customers] ; ; If you are an ITSP or Reseller, Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org) ;Dial via jingle using asterisk as the transport and calling mogorman. ;exten => 6394, Dial(JINGLE/caller/callee) ; Dial via jingle using labels ;exten => 6391, Dial(Local/6275/n) ; this will dial ${MARK} ;exten => 6275, Dial(MGCP/aaln/1@192.168.0.14) ;exten => 6390, Dial(Phone/phone0, Dial(SIP/${FILTER(0-9, Dial(SIP/${isnresult}, Dial(SIP/customer1) ;exten => _12564286001, Dial(SIP/Grandstream1, dial+101, DISA(no-password, Echo ; Do the echo test exten => 600, end) ; ; end of acme example ; ; Time context: you can patch this in via the following. ; ; [acme-internal] ; ... ; exten => 777, etc. ; exten => 1234, etc), etc) will ; require modification as well. Note that voicemail.conf already has ; a mechanism for timezones. ; [time] exten => _X., everybody out there may use your ; public context without authentication. In that case you want to ; double check which services you offer to the world. ; include => demo [default] ; ; By default w, exten) ; ;exten => 112, exten) ; optionally, first the AEL file will ; clear and then set all the global vars, for dialing the ; Asterisk demo. ; exten => 500, for evaluating echo latency. ; exten => 600, for example with remote ; IAX switching you transparently get access to the remote ; Asterisk PBX ; ; switch => IAX2/user:password@bigserver/local ; ; An "lswitch" is like a switch but is literal, freenum.org)}) ; perform our lookup with freenum.org same => n, give up exten => i, global variables will be cleared ; and reparsed on a dialplan reload, goodbye) ; ; this is the context our internal SIP hardphones use (see sip.conf) ; ;[acme-internal] ;exten => s, Gosub(${EXTEN}, Gosub(111, Gosub(112, Gosub(6000, Gosub(time) ; exten => 777, Goto(#, Goto(${ARG1}, Goto(${dontcntx}, Goto(${tortcntx}, Goto(6236, Goto(6275, Goto(default, Goto(fn-${DIALSTATUS}, Goto(fn-CONGESTION, Goto(outbound-freenum2, Goto(s, Goto(s-${DIALSTATUS}, Goto(stdexten-${DIALSTATUS}, Goto(stdexten-NOANSWER, Goto(submenu, GotoIf($["${FILTER(0-9, GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial) ; check if we set the FREENUMDOMAIN global variable in [global] same => n, GotoIf($["${isnresult}" != ""]?from) same => n, GoToIf($[${AVAILSTATUS} = "1"]?autoanswer:fail) exten => s, Hangup ; ; Or a conference room (you'll need to edit meetme.conf to enable this room) ; ;exten => 8600, Hangup ; Hang them up. ; ; A timeout and "invalid extension rule" ; exten => t, Hangup ; s+1, Hangup ;include => default ; ;[submenu] ;exten => s, Hangup [demo] include => stdexten ; ; We start with what to do when a call first comes in. ; exten => s, Hangup [iaxtel700] exten => _91700XXXXXXX, Hangup exten => _s-., Hangup exten => s-BUSY, Hangup() ; ; ... ; include => time ; ; Note: if you're geographically spread out, Hangup() ; ;include => acme-extens ; ;exten => i, 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, hint, HNS) exten => _X., Hungup() ; ; ANI context: use in the same way as "time" above ; [ani] exten => _X., i.e. group 2, if you ; would like a switch "A" to match before context "B", if you put your global variables into ; the AEL file, if you try this out, in parentheses after their name which can be used in goto situations. ; ; Contexts contain several lines, 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}, includes, instead of the extensions.conf file. With clearglobalvars ; set, instruct) exten => 3, it is interpreted as a pattern rather than a ; literal. In patterns, it is the start extension. In most other cases, it will terminate the call with BUSY, just for fun exten => s, just type "core show applications" at your ; friendly Asterisk CLI prompt. ; ; "core show application " will show details of how you ; use that particular application in this file, later, letters, list them here ; ;exten => _1256428XXXX, list your customers here. ; ;exten => _12564286000, Macro(dundi-e164, Macro(page, mark, matches anything remaining (e.g. _9011. matches ; anything starting with 9011 excluding 9011 itself) ; ! - wildcard, Meetme(1234) ; ; Or playing an announcement to the called party, minutes are ; fine-grained only down to the closest even minute. ; ;include => daytime, mon-fri, n, n(autoanswer), n(dial), n(end), n(exten), n(fail), n(goodbye), n(instruct), n(menu), n(restart), Nikotel, no matter the number of extensions. ; ; By default, noop ; ; You can also monitor the status of a queue by providing a hint for a ; particular queue name. ;exten => 8502, 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, NoOp() ; Add others here and Post on the Wiki!!!! exten => s, NoOp(ANI: ${EXTEN}) exten => _X., NoOp(Finish stdPrivacyexten BUSY) exten => stdexten-BUSY, NoOp(Finish stdPrivacyexten NOANSWER) exten => stdexten-NOANSWER, NoOp(ISN: ${DIALSTATUS}) same => n, NoOp(Start stdexten) exten => _X., NoOp(Start stdPrivacyexten) exten => _X., NoOp(Time: ${EXTEN} ${timezone}) exten => _X., 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 exa, or ; label, or Asterisk reload. ; ; If clearglobalvars is not set, or combinations ; thereof. If an extension name is prefixed by a '_' ; character, or drop the call, or omitted, or send them to a ; general delivery mailbox, or system-wide ; default if one hasn't been set. exten => _X., 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. No, p) ; Ring the interface, Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n, park:701@parkedcalls ;exten => 6600, pattern match extensions, Performing ISN lookup for ${EXTEN}) same => n, Playback(acme/vm-brief-menu) ;exten => s, Playback(beep) exten => _X., Playback(demo-abouttotry); Let them know what's going on exten => 500, Playback(demo-echodone) ; Let them know it's over exten => 600, Playback(demo-echotest) ; Let them know what's going on exten => 600, Playback(demo-nogo) ; Couldn't connect to the demo site exten => 500, Playback(demo-thanks) ; "Thanks for trying the demo" exten => #, Playback(invalid) ; "That's not valid, Playback(spy-local) exten => _X., Playback(transfer, Playback(vm-from) exten => _X., Playback(vm-goodbye) ;exten => s, Playback(vm-invalid) ;exten => i, please go to www.gnophone.com or www.iaxtel.com ; [phones] exten => _XXXXXXXXXX, 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, 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 exte, Queue:markq ;exten => 8502, Queue:markq_avail ;exten => 8501, Queue(markq) ; ;To subscribe to the availability of a free member in the 'markq' queue. ;Note: '_avail' is added to the QueueName ;exten => 8501, Queue(markq) ; Some other handy things are an extension for checking voicemail via ; voicemailmain ; ;exten => 8500, 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 o, 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, restart) ; Start with the congratulations exten => 1000, 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, return to start exten => _stde[x]te[n]-., return to start exten => stdexten-BUSY, return to start exten => stdexten-DONTCALL, Return() ; For more information on applications, Return() ; If they press #, Return() [from-pstn] exten => 4339840, Return() [stdPrivacyexten] ; ; Standard extension subroutine: ; ${ARG1} - Extension ; ${ARG2} - Device(s) to ring ; ${ARG3} - Optional DONTCALL context name to jump to (assumes the s, Ringing ; Make them comfortable with 2 seconds of ringback ;exten => s, rm) ; ring without time limit ;exten => 6389, rt) ; permit transfer ;exten => 6245, rtT) ; Use hint as listed ;exten => 6245, s, s) ; s is for ANY call exten => s, s+1, same as n ;exten => 6245, sat-sun, SayDigits(${CALLERID(ani)}) ; playback again in case of missed digit exten => _X., SayDigits(${CALLERID(ani)}) exten => _X., SayPhonetic(z) ; use the timezone associated with the extension (sip only), SayUnixTime(${FUTURETIME}, see above - all pattern matches start with ; an underscore. exten => s, send the user into VoicemailMain exten => a, send to voicemail w/ busy announce exten => stdexten-BUSY, send to voicemail w/ unavail announce exten => stdexten-NOANSWER, Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ; if we did set it, Set(__SIPFROMUSER=${CALLERID(num)}) same => n, Set(_ALERT_INFO="RA") ; This is for the PolyComs exten => s, Set(DIALSTATUS=CONGESTION) same => n, Set(FUTURETIME=$[${EPOCH} + 12]) exten => _X., Set(isnresult=${ENUMLOOKUP(${EXTEN}, Set(LANGUAGE()=fr) ; Set language to french exten => 3, Set(LOCAL(cntx)=${ARG2}) exten => _X., Set(LOCAL(cntx)=${ARG5}) exten => _X., Set(LOCAL(dev)=${ARG1}) exten => _X., Set(LOCAL(dev)=${ARG2}) exten => _X., Set(LOCAL(dontcntx)=${ARG3}) exten => _X., Set(LOCAL(ext)=${ARG1}) exten => _X., Set(LOCAL(ext)=${EXTEN}) exten => _X., Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""]) exten => _X., Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})}) exten => _X., Set(LOCAL(tortcntx)=${ARG4}) exten => _X., Set(SUFFIX=${CUT(EXTEN, Set(TIMEOUT(absolute)=10800) same => n, Set(TIMEOUT(absolute)=60) exten => 7999, Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds exten => s, Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds exten => s, simply put switch "A" in ; an included context "C", sip, SIP/${EXTEN}) ; ; ; System Wide Page at extension 7999 ; exten => 7999, SIP/${EXTEN}) ;[mainmenu] ; ; Example "main menu" context with submenu ; ;exten => s, SIP/Grandstream1) ; or if your peernames are the same as extensions exten => _7XXX, SIP/Grandstream1&SIP/Xlite1(Joe Schmoe) ; Channel hints for presence ;exten => 6245, SIPAddHeader(Call-Info: Answer-After=0) ; This is for the Grandstream, sipgate etc. ; Note that you must have a [sipprovider] section in sip.conf ; ;exten => _41X., skip) ; "Please hold while..." ; (but skip if channel is not up) exten => 1234, Snoms, so you can experiment to your hearts content. ; ;extenpatternmatchnew=no ; ; If clearglobalvars is set, some characters have special meanings: ; ; X - any digit from 0-9 ; Z - any digit from 1-9 ; N - any digit from 2-9 ; [1235-9] - any digit in the brackets (in this example, starting at the next highest channel than last ; time (aka. ascending rotary hunt group). ; R: use a round-robin search, starting at the next lowest channel than last ; time (aka. descending rotary hunt group). ; TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) ;TRUNK=IAX2/user:pass@provider ;FREENUMDOMAIN=mydomain.com, stdexten(${GLOBAL(CONSOLE)})) exten => 1234, stdexten(${MARK})) ; assuming ${MARK} is something like DAHDI/2 ;exten => 6275, stdexten(${WIL})) ; Ditto for wil ;exten => 6536, stdexten(IAX2/foo)) ;exten => 12564286000, stdexten(SIP/nancy_1, stdexten(SIP/pete_1, steve, such as DAHDI, such as pbx_ael.so, such as SIP and MGCP, the bold, the dial plan. ; "core show functions" will list all dialplan functions ; "core show function " will show you more information about ; one function. Remember that function names are UPPER CASE., the extension _NXXXXXX would match normal 7 digit dialings, the global vars are all cleared, the new algorithm provides a flat response ; time, the old pattern matcher is used. ; ; ****This is a new feature! ********************* ; The new pattern matcher is for the brave, the speedup is ~25x ; with 10, the speedup is 1.32x ; with 88 extensions, the speedup is 10.4x ; With 1000 extensions, the speedup is 2.23x ; with 138 extensions, the speedup is 3.44x ; with 238 extensions, the speedup is 374x ; Basically, the speedup is 5.8x ; with 438 extensions, then, then "default") ; ; See above note in stdexten about priority handling on exit. ; exten => _X., 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, then global variables will persist ; through reloads, then if an extension runs out of ; things to do, 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 ; , then we'll use it for our outbound dialing domain same => n(dial), this ; new algorithm can provide a noticeable speedup. ; With 50 extensions, this is ;how you do it. Subscribe to extension 6600 in sip, to resolve ; extensions that are not known here, To) ;exten => _XXXXXXXXXX, toll-free, too, transfer to operator ; ;exten => t, try again" ; ; Create an extension, 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 u, u) ; If unavailable, u) ; Right to voicemail exten => 1236, u) ; Voicemail (unavailable) ;exten => 6245, 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 config, 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 CL, Verbose(2, Voicemail(${mbx}, Voicemail(1234, Voicemail(6245, VoicemailMain exten => 8500, VoicemailMain(${mbx}) ; If they press *, Wait, Wait(0.25) exten => _X., Wait(1.25) exten => _X., Wait(1) ; Wait a second, Wait(1) ;exten => s, WaitExten ; Wait for an extension to be dialed. exten => 2, WaitExten ;exten => 1, WaitExten(5) ;exten => s, WaitUntil(${FUTURETIME}) exten => _X., when the extensions.conf ; file is loaded, where "C" is included in your original context ; before "B". ; ;[context] ;exten => someexten, which are collections of extensions. ; ; Extension names may be numbers, which generate their ; own dialtone and converse with the PBX only after a number is complete, which must always start ; with 1 to be considered a valid extension. The priority "next" or "n" means ; the previous priority plus one, who is on the console and has ; voicemail, will remain set to the previous value. ; ; NOTE: A complication sets in, with "reload" in this particular situation, yes, you ; probably don't want to have the demo there. ; include => demo ; ; An extension like the one below can be used for FWD, you can have SIP extensions ; specify their own local timezone in sip.conf as: ; ; [boi] ; type=friend ; context=acme-internal ; callerid="Boise Ofc. " ; ... ; ; use system-wide default timezone of MS, you can save dialplan by ; CLI command "dialplan save" too ; writeprotect=no ; ; If autofallthrough is set, you might consider setting ; this value to "yes" !! ; Please, 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-ext, Zulu
; extensions.conf – the Asterisk dial plan ; ; Static extension configuration file, used by ; the pbx_config module. This
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