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Asterisk info Tutorial

Adding Listen, Whisper, and Barge to FreePBX or Asterisk

July 8, 2013February 12, 2016 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

Asterisk info Tutorial

Allow Asterisk Manager Access via HTTP

June 28, 2013June 28, 2013 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”

Asterisk info Tutorial

Asterisk Auto Recording even Manual Call on Phone using Monitor

June 28, 2013June 28, 2013 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 =>

Customizing the Music On Hold (MOH) message on Asterisk

June 28, 2013 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

Asterisk info Tutorial

Adding Listen, Whisper, and Barge to FreePBX or Asterisk

June 28, 2013February 24, 2016 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

Asterisk GOautodial info Website

Configuring Digium cards with Asterisk , goautodial , vicidial , vicidialnow , freepbx

June 21, 2013February 24, 2016 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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Google talk Tips and tricks

April 23, 2013February 12, 2016 eduguru 0 Comments Accounts, and enter your custom name. As a result all of your emails will have that nick as well, and then Edit info. Click on the second radio button, b=2, but forces it to start no matter what option was set. * /S upgrade: Used when upgrading Google Talk * /register: registers Google Talk in the registry, c=3 .... When you translated this message it said: 'play wumpus.game'. To play this game you had to invite wumpus.game@gmail.com as a friend. wumpus.game@gmail.com is always offline now. You can still, etc.. through Google Talk. Unplug your microphone. Double click on the speaker icon in the lower right corner. This will open up "Volume Control". Select "Options" and then "Properties". Then check th, Google talk Tips and tricks, Google talk Tips and tricks gtalk * CTRL + Mousewheel up/down over input textbox: Change the font size of the textbox. * F9: Open Gmail to send an email to your friend * F11: Start a call with your fr, includig the GMail Compose method. * /checkupdate: check for newer versions * /plaintextauth: uses plain authentication mechanism instead then Google's GAIA mechanism. Used for testing the plain metho, it will be clickable when someone opens a conversation window with you. Contacts * You don’t need to say Yes or No when someone wants to add you as a friend; you can simply ignore it, it will check the registry settings to see if it needs to be started or not. If the "Start automatically with Windows" option is unchecked, it won't start. * /forcestart: same as /autostart, MP3, or in the tray) Conversation Text * A message can be 32767 characters long. * Certain smileys are recognized by Google Talk and will be shown in blue. :-| :-O :-x :-P :-D ;-) :-( :| :O :x :P :D :) :( , put it between _underscores_ * You can insert special characters like ♥♫☺ with 'Start / Programs / Accessories / System Tools / Character Maps'. Conversation Window * Drag a conversation window on top, put it between *asteriks* * To write text in italic, send off a message to Google. * There was a hidden game in Google Talk. In the about screen you could see 'play 23 21 13 16 21 19 . 7 1 13 5'. Each number represented a letter. a=1, someone with whom you chat often will automatically turn to be your friend, the request will go away. (On the other hand, there are no other known GAIA servers. * /mailto email@host.com: send an email with Gmail * /diag: start Google Talk in diagnostic mode * /log: probably has something to do with the diagnostic logging, there is no way to seperate the two. * You can add a website in your custom message, unless you disable this). * The Gmail account 'user@gmail.com' can't be invited as your friend. Sound & Video * It's possible to broadcast music, you can right click it to close it without focusing the conversation window. Conference Calls * What you need to do to have conference calls: Open up a copy of Google Talk on all computers with which

Google talk Tips and tricks * CTRL + Mousewheel up/down over input textbox: Change the font size of the textbox.