Asterisk SIP Trunk calling thorugh other server – Communicating two asterisk server
Asterisk SIP Trunk calling thorugh other server – Communicating two asterisk server Configuration:- Configure the following sip account in both
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copy a folder under linux server to windows server
eduguru 1 Comment copy a folder under linux server to windows server, credentials=/etc/sambapasswords 0 0 Append an entry to your crontab like this, dmask=0775, fmask=0664, gid=users, if you need to do a backup daily at 1AM: 0 1 * * * cp /path/to/yourbackup /mnt/win FTP solution: you may install a ftp server on your windows machine. filezilla server do perfectly the job. setup a ft, iocharset=iso8859-15, MS-Windows shared folder: You can share data between windows and linux system for such use : For example you would like to access MS-Windows share called //windowsserver/sharename by mounting to /mnt/, password=mypassword //windowsserver/sharename /mnt/win Next create the password file /etc/sambapasswords: cat > /etc/sambapasswords username = winntuser password = mypassword make sure that only ro
MS-Windows shared folder: You can share data between windows and linux system for such use : For example you would
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Asterisk Full RealTime Database example
eduguru 2 Comments `accountcode` varchar(20) default NULL, `adminpin`, `adminpin` varchar(20) default NULL, `allow` varchar(100) default 'g729;ilbc;gsm;ulaw;alaw', `amaflags` varchar(7) default NULL, `announce` varchar(128) default NULL, `announce_frequency` int(11) default NULL, `announce_holdtime` varchar(128) default NULL, `announce_round_seconds` int(11) default NULL, `app`, `app` varchar(20) NOT NULL default '', `appdata` varchar(128) NOT NULL default '', `appdata`) values ('from-sip', `attach` varchar(4) NOT NULL default 'yes', `callback` varchar(10) NOT NULL, `callerid` varchar(80) default NULL, `callgroup` varchar(10) default NULL, `cancallforward` char(3) default 'yes', `canreinvite` char(3) default 'yes', `context` varchar(128) default NULL, `context` varchar(20) NOT NULL default '', `context` varchar(50) NOT NULL, `context` varchar(80) default NULL, `context`) ) ENGINE=MyISAM AUTO_INCREMENT=2001 DEFAULT CHARSET=latin1; CREATE TABLE `queue_table` ( `name` varchar(128) NOT NULL, `context`) VALUES ('my_queue', `customer_id` varchar(11) NOT NULL default '0', `defaultip` varchar(15) default NULL, `defaultuser` varchar(80) NOT NULL, `delete` varchar(4) NOT NULL default 'no', `deny` varchar(95) default NULL, `dialout` varchar(10) NOT NULL, `disallow` varchar(100) default 'all', `domain`, `domain` varchar(128) NOT NULL default '', `dtmfmode` varchar(7) default NULL, `email` varchar(50) NOT NULL, `envelope` varchar(4) NOT NULL default 'no', `eventmemberstatus` tinyint(1) default NULL, `eventwhencalled` tinyint(1) default NULL, `exten` varchar(20) NOT NULL default '', `forcegreetings` varchar(4) NOT NULL default 'no', `forcename` varchar(4) NOT NULL default 'no', `fromdomain` varchar(80) default NULL, `fromuser` varchar(80) default NULL, `fullcontact` varchar(80) default NULL, `fullname` varchar(150) NOT NULL, `hidefromdir` varchar(4) NOT NULL default 'yes', `host` varchar(31) NOT NULL, `insecure` varchar(4) default NULL, `interface`, `interface` varchar(128) default NULL, `interface`) ) ENGINE=MyISAM AUTO_INCREMENT=6 DEFAULT CHARSET=latin1; CREATE TABLE `meetme` ( `confno` varchar(80) NOT NULL default '0', `ipaddr` varchar(15) NOT NULL, `joinempty` varchar(128) default NULL, `language` char(2) default NULL, `lastms` int(11) NOT NULL, `leavewhenempty` varchar(128) default NULL, `mailbox` varchar(11) NOT NULL default '0', `mailbox` varchar(50) default NULL, `mask` varchar(95) default NULL, `maxlen` int(11) default NULL, `md5secret` varchar(80) default NULL, `memberdelay` int(11) default NULL, `membername`, `membername` varchar(40) default NULL, `members` int(11) NOT NULL default '0', `members`) VALUES ('my_conf', `monitor_format` varchar(128) default NULL, `monitor_join` tinyint(1) default NULL, `musiconhold` varchar(100) default NULL, `musiconhold` varchar(128) default NULL, `name` varchar(80) NOT NULL, `nat` varchar(5) NOT NULL default 'no', `nextaftercmd` varchar(4) NOT NULL default 'yes', `operator` varchar(4) NOT NULL default 'no', `pager` varchar(50) NOT NULL, `password` varchar(5) NOT NULL default '0', `paused` int(11) default NULL, `paused`) VALUES (1, `paused`) VALUES (2, `penalty`, `penalty` int(11) default NULL, `periodic_announce` varchar(50) default NULL, `periodic_announce_frequency` int(11) default NULL, `permit` varchar(95) default NULL, `pickupgroup` varchar(10) default NULL, `pin`, `pin` varchar(20) default NULL, `port` varchar(5) NOT NULL, `priority` tinyint(4) NOT NULL default '0', `priority`), `qualify` char(3) default NULL, `queue_callswaiting` varchar(128) default NULL, `queue_holdtime` varchar(128) default NULL, `queue_lessthan` varchar(128) default NULL, `queue_minutes` varchar(128) default NULL, `queue_name`, `queue_name` varchar(128) default NULL, `queue_reporthold` varchar(128) default NULL, `queue_seconds` varchar(128) default NULL, `queue_thankyou` varchar(128) default NULL, `queue_thereare` varchar(128) default NULL, `queue_youarenext` varchar(128) default NULL, `regexten` varchar(80) NOT NULL, `regseconds` int(11) NOT NULL default '0', `regserver` varchar(100) default NULL, `reportholdtime` tinyint(1) default NULL, `restrictcid` char(1) default NULL, `retry` int(11) default NULL, `review` varchar(4) NOT NULL default 'no', `ringinuse` tinyint(1) default NULL, `rtpholdtimeout` char(3) default NULL, `rtptimeout` char(3) default NULL, `saycid` varchar(4) NOT NULL default 'yes', `sayduration` varchar(4) NOT NULL default 'no', `saydurationm` tinyint(4) NOT NULL default '1', `secret` varchar(80) default NULL, `sendvoicemail` varchar(4) NOT NULL default 'no', `servicelevel` int(11) default NULL, `setinterfacevar` tinyint(1) default NULL, `stamp` timestamp NOT NULL default CURRENT_TIMESTAMP on update CURRENT_TIMESTAMP, `strategy` varchar(128) default NULL, `timeout` int(11) default NULL, `timeoutrestart` tinyint(1) default NULL, `type` varchar(6) NOT NULL default 'friend', `tz` varchar(10) NOT NULL default 'central', `useragent` char(255) default NULL, `username`, `username` varchar(64) NOT NULL default '', `weight` int(11) default NULL, `wrapuptime` int(11) default NULL, '_*0', '_9X.', '_X.', '', ''), '${CALLERID(num)}@from-sip'), '${EXTEN}@from-sip'), '1000', '11111111', '12121111111', '12122222222', '12127777777', '12129999999', '2000', '4444', '5555', 'Dial', 'dynamic', 'friend'); ; INSERT INTO asteriskrealtime.sip_buddies ( NAME, 'friend'); inster the dial-plan in the extensions table insert into `extensions`(`context`, 'from-sip', 'from-sip'); Insert the two extensions 1000 and 2000 as members: INSERT INTO `queue_member_table` (`uniqueid`, 'MeetMe', 'my_conf'); add voicemail to ext 1000 & 2000: INSERT INTO asteriskrealtime.voicemail_users (customer_id, 'my_queue'), 'no', 'Queue', 'SIP/${EXTEN:1}@deltathree'), 'SIP/${EXTEN}|30'), 'SIP/1000', 'SIP/1000@from-sip', 'SIP/1000|60'), 'SIP/2000', 'SIP/2000@from-sip', 'SIP/2000|60'), 'sipauth.deltathree.com', 'User-A', 'User-B', 'UserA@myemail.com'); ; INSERT INTO asteriskrealtime.voicemail_users (customer_id, 'UserB@myemail.com'); Insert the new queue "my_queue": INSERT INTO `queue_table` (`name`, 'VoiceMail', 'VoiceMailMain', ${EXTEN}, 0); ; INSERT INTO `queue_member_table` (`uniqueid`, 0); insert the conference bridge: INSERT INTO `meetme` (`confno`, 0); Restart asterisk and go into the CLI: asterisk -rx "restart now" asterisk -r Check the asterisk realtime status localhost*CLI> realtime mysql status general connected to asteriskrealtime@127.0., 1, 1234, 2, 3, 4, 5 are "off" turn to "on" chkconfig --level 345 mysqld on set your mysql root password (in needed): mysqladmin -u root password password go into the mysql CLI: mysql -u root -p insert the "password" Cr, and start it if needed: /etc/init.d/mysqld status /etc/init.d/mysqld start in order to start mysql after reboot check the status chkconfig | grep mysql If 3, Asterisk Full RealTime Database example, change the extconfig.conf file as follow [settings] sippeers => mysql, Configure asterisk MySql connection: create the res_mysql.conf file in /etc/asterisk vi /etc/asterisk/res_mysql.conf enter the following: [general] dbhost = 127.0.0.1 dbname = asteriskrealtime dbuser , context, defaultuser, email ) VALUES ('1000', email ) VALUES ('2000', extensions voicemail => mysql, fullname, general, Hangup, HOST, KEY `id` (`id`) ) ENGINE=MyISAM AUTO_INCREMENT=257 DEFAULT CHARSET=latin1; CREATE TABLE `voicemail_users` ( `uniqueid` int(11) NOT NULL auto_increment, KEY `mailbox_context` (`mailbox`, KEY `name_2` (`name`) ) ENGINE=MyISAM AUTO_INCREMENT=893 DEFAULT CHARSET=latin1; CREATE TABLE `extensions` ( `id` int(11) NOT NULL auto_increment, mailbox, meetme Configure sip.conf and exensioncs.conf sip.conf should Register your account and DID with the SIP provider Change the default context to "from-sip" for the inbound calls from the SIP provider. , nat, NULL, PASSWORD, port 3306 with username root for 12 seconds., PRIMARY KEY (`confno`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1; verify that all tables created: mysql> show tables; +----------------------------+ | Tables_in_asteriskrealtime | +-------------------, PRIMARY KEY (`context`, PRIMARY KEY (`id`), PRIMARY KEY (`name`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1; CREATE TABLE `queue_member_table` ( `uniqueid` int(10) unsigned NOT NULL auto_increment, PRIMARY KEY (`uniqueid`), qualify, queue_member_table meetme => mysql, queue_table queue_members => mysql, secret, sip_buddies extensions => mysql, TYPE) VALUES ( '1000', TYPE) VALUES ( '2000', TYPE) VALUES ( 'deltathree', UNIQUE KEY `name` (`name`), UNIQUE KEY `queue_interface` (`queue_name`, voicemail_users queues => mysql, we we call it "asteriskrealtime" create database if not exists `asteriskrealtime`; change to the asteriskrealtime database: use asteriskrealtime; now we will create the following tables: sip_buddies e, we will use a very short files instead in order to simplefy the setup: cp /etc/asterisk/extensions.conf /etc/asterisk/extensions.conf_orig cp /etc/asterisk/extconfig.conf /etc/asterisk/extconfig.conf_, yes
Configure asterisk MySql connection: create the res_mysql.conf file in /etc/asterisk vi /etc/asterisk/res_mysql.conf enter the following: [general] dbhost = 127.0.0.1 dbname
Read moreinstall phpmyadmin on centos 6
eduguru 0 Comments 32 Bit (i386): # rpm -Uvh http://dl.fedoraproject.org/pub/epel/5/i386/epel-release-5-4.noarch.rpm CentOS/RHEL 5, 32 Bit (i386): # rpm -Uvh http://download.fedoraproject.org/pub/epel/6/i386/epel-release-6-8.noarch.rpm CentOS/RHEL 6, 64 Bit (x86_64): # rpm -Uvh http://dl.fedoraproject.org/pub/epel/5/x86_64/epel-release-5-4.noarch.rpm Step 2: Install Package using Yum Install phpMyAdmin package using yum command line tool. # yum in, 64 Bit x86_64): # rpm -Uvh http://download.fedoraproject.org/pub/epel/6/x86_64/epel-release-6-8.noarch.rpm CentOS/RHEL 5, Install package using one of below urls. CentOS/RHEL 6, install phpmyadmin on centos 6, refresh-packagekit, security Loading mirror speeds from cached hostfile * base: mirror.nbrc.ac.in * epel: epel.mirror.net.in * extras: mirror.nbrc.ac.in * updates: mirror.nbrc.ac.in Setting up Install Process Resolving D, Step 1: Add EPEL Repository in System First step is to install EPEL repository in your system
Step 1: Add EPEL Repository in System First step is to install EPEL repository in your system, Install package using
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Install MySQL on CentOS 6.4 – Asterisk mysql database configuration
eduguru 1 Comment Install MySQL on CentOS 6.4 - Asterisk mysql database configuration, nstall MySQL on CentOS 6.4 Install MySQL on CentOS 6.4 su to root install mysql and mysql-server download and install all dependencies done start mysql
How to Install MySQL on CentOS 6.4 su to root install mysql and mysql-server download and install all dependencies done
Read moreAdding 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
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