Install MySQL on CentOS 6.4 – Asterisk mysql database configuration
How to Install MySQL on CentOS 6.4 su to root install mysql and mysql-server download and install all dependencies done
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Asterisk
Adding Listen, Whisper, and Barge to FreePBX or Asterisk
eduguru 0 Comments ) exten => _*222x.#, ) exten => _*223x.#, ) exten => _*224x.#, 1, Adding Listen, and all parties can hear each other. Be warned, and Barge to FreePBX or Asterisk, and Barge to FreePBX or Asterisk ;listen exten => _*222x.#, and the spied-on channel (agent) can also hear the manager, and when comfortable, Answer exten => _*222x.#, Answer exten => _*223x.#, Answer exten => _*224x.#, but not the bridged channel, but they cannot hear the manager. Whisper: Whisper to the agent. The manager can hear both the spied-on and bridged channels, ChanSpy(sip/${EXTEN:4}, Hangup, Hangup ;barge exten => _*224x.#, Hangup ;whisper exten => _*223x.#, hence “whisper.” Barge: Barge in on both channels. The manager channel is joined onto the spied-on and bridged channel, if the original agent leaves the call, initiate a 3way call to your extension so you can continue the call without the agent., Listen: Monitor an agents call. The manager can hear both the spied-on and bridged channels, Macro(user-callerid, n, NoCDR exten => _*222x.#, NoCDR exten => _*223x.#, NoCDR exten => _*224x.#, q) exten => _*222x.#, qB) exten => _*224x.#, qw) exten => _*223x.#, the call is dropped. This is not a 3-way call. (However you can barge in, Wait(1) exten => _*222x.#, Wait(1) exten => _*223x.#, Wait(1) exten => _*224x.#, Whisper
Adding Listen, Whisper, and Barge to FreePBX or Asterisk Listen: Monitor an agents call. The manager can hear both the
Read moreAllow Asterisk Manager Access via HTTP
eduguru 0 Comments "astman" like interface. You can login with your username/secret for manager and have a basic view of channels as well as transfer and hangup calls. It's only tested in Firefox, Allow Asterisk Manager Access via HTTP, Asterisk stores a cookie on your browser (valid for the length of httptimeout) which is used to connect to the same session. http://localhost:8088/asterisk/rawman?action=status Assuming you've already, Asterisk will serve up a demo page which presents a live, but could probably be made to run in other browsers as well., but very basic, Make sure you have both "enabled = yes" and "webenabled = yes" setup in /etc/asterisk/manager.conf You may also use "httptimeout" to set a default timeout for HTTP connections. Make sure you have a ma, theoretically compatible with RICO http://localhost:8088/asterisk/static/ajamdemo.html If you have enabled static content support and have done a make install, this URI will give you a "raw" manager output for the "status" command. http://localhost:8088/asterisk/mxml?action=status This will give you the same status view but represented as AJAX data
Make sure you have both “enabled = yes” and “webenabled = yes” setup in /etc/asterisk/manager.conf You may also use “httptimeout”
Read moreAsterisk Auto Recording even Manual Call on Phone using Monitor
eduguru 1 Comment %Y%m%d-%H%M%S)}) exten => _011., %Y%m%d-%H%M%S)}) exten => _1NXXNXXXXXX, ${CALLFILENAME}, 1, 103, 2, 3, 4, 5, AGI(agi://127.0.0.1:4577/call_log) exten => _011., AGI(agi://127.0.0.1:4577/call_log) exten => _1NXXNXXXXXX, all calls recording asterisk, Asterisk Auto Recording even Manual Call on Phone using Monitor, asterisk calls recording, CONGESTION, Congestion exten => _011., Congestion exten => _1NXXNXXXXXX, Dial(${SIPTRUNK}/${EXTEN}, exten => _1NXXNXXXXXX, m) exten => _011., m) exten => _1NXXNXXXXXX, Monitor(wav, Set(CALLFILENAME=${CALLERID(all)}-${EXTEN}-${STRFTIME(${EPOCH}, tTor) exten => _011., tTor) exten => _1NXXNXXXXXX
exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _1NXXNXXXXXX,2,Set(CALLFILENAME=${CALLERID(all)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) exten => _1NXXNXXXXXX,3,Monitor(wav,${CALLFILENAME},m) exten => _1NXXNXXXXXX,4,Dial(${SIPTRUNK}/${EXTEN},,tTor) exten => _1NXXNXXXXXX,5,Congestion exten => _1NXXNXXXXXX,103,Congestion exten =>
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(${FUTURETIME}) exten => _X., when the extensions.conf ; file is loaded, where "C" is included in your original context ; before "B". ; ;[context] ;exten => someexten, which are collections of extensions. ; ; Extension names may be numbers, which generate their ; own dialtone and converse with the PBX only after a number is complete, which must always start ; with 1 to be considered a valid extension. The priority "next" or "n" means ; the previous priority plus one, who is on the console and has ; voicemail, will remain set to the previous value. ; ; NOTE: A complication sets in, with "reload" in this particular situation, yes, you ; probably don't want to have the demo there. ; include => demo ; ; An extension like the one below can be used for FWD, you can have SIP extensions ; specify their own local timezone in sip.conf as: ; ; [boi] ; type=friend ; context=acme-internal ; callerid="Boise Ofc. " ; ... ; ; use system-wide default timezone of MS, you can save dialplan by ; CLI command "dialplan save" too ; writeprotect=no ; ; If autofallthrough is set, you might consider setting ; this value to "yes" !! ; Please, you will need to ; activate them within asterisk.conf with the "execincludes" option. They ; are otherwise considered a security risk. ;#exec /opt/bin/build-extra-contexts.sh ;#exec /opt/bin/build-ext, Zulu
; extensions.conf – the Asterisk dial plan ; ; Static extension configuration file, used by ; the pbx_config module. This
Read moreCustomizing the Music On Hold (MOH) message on Asterisk
eduguru 0 Comments Customizing the Music On Hold (MOH) message on Asterisk, enter: for f in `ls *.mp3` ; do FILE=$(basename $f .mp3) ; ffmpeg -i $FILE.mp3 -ar 8000 -ac 1 -ab 64 $FILE.wav -ar 8000 -ac 1 -ab 64 -f mulaw $FILE.pcm -map 0:0 -map 0:0 ; done, execute: ffmpeg -i "[input file]" -ar 8000 -ac 1 -acodec pcm_s16le -f s16le "[output file].sln" To convert the existing mp3 files to mono wav and ulaw pcm, for Asterisk to be able to convert immediately. For instance, g722, g729, gsm and in Asterisk Native SLN, incoming calls on your standard lines are handled and the greeting message is played ( the_welcome_ class being specified for the queue welcome); let us see how it is defined. MOH is described in /etc, this file contains this: [general] [default] mode=files directory=moh You will need to append the following section to it: [welcome] mode=files sort=alpha directory=/etc/asterisk/moh1 By this record, to convert to Native SLN with ffmpeg, ulaw, you should first convert the message with ffmpeg@ (the package media-video/ffmpeg is pre-installed in Calculate Linux Desktop) to the most commonly used codecs: _alaw, you tell the Asterisk server to play files from the directory /etc/asterisk/moh1 by alphabetic order. The recorded message is put in this directory. To decrease server load
incoming calls on your standard lines are handled and the greeting message is played ( the_welcome_ class being specified for
Read moreAdding Listen, Whisper, and Barge to FreePBX or Asterisk
eduguru 0 Comments Adding Listen, agent who is the spied-on channel, also known as joining multiple calls, and all parties can hear each other. Be warned, and barge as follows: Listen: Monitor an agents call. The manager can hear both the spied-on and bridged channels, and Barge to FreePBX or Asterisk, and caller who is the bridged channel. We define listen, and the spied-on channel (agent) can also hear the manager, and when comfortable, but not the bridged channel, but they cannot hear the manager. Whisper: Whisper to the agent. The manager can hear both the spied-on and bridged channels, hence “whisper.” Barge: Barge in on both channels. The manager channel is joined onto the spied-on and bridged channel, if the original agent leaves the call, If you are running a call center on FreePBX or Asterisk, initiate a 3way call to your extension so you can continue the call without the agent., most likely you will want the ability to listen in on agents calls, or connected two calls to a manager, or other variations of barging in on a bridged channel. For the purpose of this article, the call is dropped. This is not a 3-way call. (However you can barge in, we will define manager as the callee who is the spying channel, Whisper
If you are running a call center on FreePBX or Asterisk, most likely you will want the ability to listen
Read moreConfiguring Digium cards with Asterisk , goautodial , vicidial , vicidialnow , freepbx
eduguru 1 Comment ) "if goautodial/vicidial/vicibox use the below dialplan in carrier or custome dialplan" exten => _9XXXXX., ) then the Dhadi driver will be by default installed, 1, 2, 3, AGI(agi://127.0.0.1:4577/call_log) exten => _9XXXXX., Configuring Digium cards with Asterisk, Configuring Digium cards with vicidial, Dial(DAHDI/g0/${EXTEN:1}, Dial(DAHDI/g0/${EXTEN}, elastix, follow the below steps type dahdi_genconf -vvvvvvv -this will auto install configure the digium card driver and conf files [root@localhost ~]# dahdi_genconf -vvvvvv Default parameters from /etc/dahdi/, freepbx, goautodial, Hangup 8a. if your using freepbx based pbx then no need to above line, How to configure the Digium PRI cards in Asterisk or vicidial or Goautodial or vicidialnow or Freepbx If you are using the precompiled iso of asteirsk software like ( trixbox, if digium card is recognised it shows output as below 0e:08.0 Ethernet controller: Digium, if not you need to install the dahdi driver manually.(installation of dahdi explained at last of this doc) steps to configure the Digium cards First you need to check whether card is in E1 or T1 mode., if T1 then it should be open. then insert the card into the server. Access your asterisk server via ssh, if your PRI line is E1 then you need close the jumper, Inc. Wildcard TE121 single-span T1/E1/J1 card (PCI-Express) (rev 11) Once the card is recognised, just you need to add g0 in the DAHDI identifier under trunk (add new dahdi trunk) 9. And the default incomming context will the from-pstn Installing the DAHDI driver a. you need to install libpri if y, pbxinaflash, putty will be good tool to connect the server remotely via ssh First you need to check whether the card is recognised by the your server type lspci this will output pci boards connected in the server, so in your dialplan you need to mention g0 8. so you need write dialplan as exten => _X., some small changes to be done to enable the channels in the asterisk go to vi /etc/asterisk/chan_dahdi.conf go to last line of the file by pressing shift+g then add this line #include dahdi-channels.c, TtoR) exten => _9XXXXX., vicibox, vicidial, vicidialnow
How to configure the Digium PRI cards in Asterisk or vicidial or Goautodial or vicidialnow or Freepbx If you
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Asterisk GUI
eduguru 0 Comments $PORT is bindport from http.conf, agent, and $PREFIX is the prefix from http.conf, and feature requests may be submitted at http://issues.asterisk.org and should reference the Asterisk GUI project. You may also find peer support in the #asterisk-gui IRC channel and the Asterisk GUI , and it can be omitted if blank. http://$IP:$PORT/$PREFIX/static/config/index.html Check on the Asterisk CLI that Asterisk is receiving the values you've set. http show status manager show settings Out, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details. Please contact Digium for infor, call, command, config, Inc. This program is distributed in the hope that it will be useful, Inc. This program is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation, Introduction to Asterisk GUI Asterisk GUI is a framework for the creation of graphical interfaces for configuring Asterisk. Download While package release is inconsistent and infrequent, log, originate Access Access Asterisk GUI via a URL formatted in the following way, originate write = system, patches, read, specifically: http.conf: Enable http. [general] enabled=yes enablestatic=yes #bindaddr=0.0.0.0 # allow GUI to be accessible from all IP addresses. bindaddr=127.0.0.1 # require access from the machine , though bugs, verbose, version 2 only. This software is also available under commercial terms from Digium, where $IP is the IP address on which both Asterisk and Asterisk GUI are installed, write, you can always get a current copy of Asterisk GUI from subversion. The current stable version will always be under branches and is currently located in branches/2.0. Support Please note that Asterisk
Introduction to Asterisk GUI Asterisk GUI is a framework for the creation of graphical interfaces for configuring Asterisk. Download While
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