Asterisk

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

Asterisk

Asterisk GUI

April 12, 2013March 30, 2016 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

Asterisk

Calendaring Dialplan Asterisk Calendaring

April 11, 2013February 16, 2015 eduguru 0 Comments ${EPOCH}, ${EPOCH})}) same => n, 1, 1) same => n(closed), A common business PBX scenario is would be executing dialplan logic based on when the business is open and the phones staffed. If the business is closed for holidays, Answer same => n, Calendaring Dialplan Asterisk Calendaring, description)}) same => n, GotoIf(${CALENDAR_BUSY(officehours)}?closed:attendant, Hangup, it is sometimes desirable to play a message to the caller stating why the business is closed. The standard way to do this in asterisk has been doing a series of GotoIfTime statements or time-based inc, Playback($[${ISNULL(soundfile)} ? generic-closed :: ${soundfile}]) same => n, s, Set(id=${CALENDAR_QUERY(office, Set(soundfile=${CALENDAR_QUERY_RESULT(${id}, the adminstrator only needs to set up a calendar that contains the various holidays or even recurring events specifying the office hours. A custom greeting filename could even be contained in the desc, xten => 011123456

A common business PBX scenario is would be executing dialplan logic based on when the business is open and the

Asterisk

Open source Voice Broadcasting Software (Dialer)

April 1, 2013May 3, 2016 eduguru 0 Comments Ge a Call center dialer to voice broadcasting. For more info: Mail us: info@eduguru.in Visit Us: http://intextech.com, Open source Voice Broadcasting Software (Dialer), Voice Broadcasting, voicebroad casting, What is Voice Broadcasting Voice broadcasting is a mass communication technique

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ADS Asterisk

Dialer for small and medium organisation – Call center Dialer

April 1, 2013February 24, 2016 eduguru 0 Comments Dialer for small and medium organisation - Call center Dialer, Dialer for small and medium organisation - Call center Dialer Get a free open source dialer with all the advance feature list. iNet Expert TechnologyContact Us: info@eduguru.in Visit Us :http://inetex

Dialer for small and medium organisation – Call center Dialer Get a free open source dialer with all the advance

Asterisk

March 26, 2013February 24, 2016 eduguru 2 Comments CRM support, Voice and Data for both Domestic and International Callcenter. For any query mail us info@eduguru.in iNet Expert Technology Visit Us : http://inetextech.com/ For any query mail us info@eduguru.in, We provide callcenter technology software like Dialer with life time free licence with all the advanced features for inbound and outbound calling

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Asterisk

Customizing the Music On Hold (MOH) message on Asterisk

Adding Listen, Whisper, and Barge to FreePBX or Asterisk

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

Asterisk GUI

Calendaring Dialplan Asterisk Calendaring

Open source Voice Broadcasting Software (Dialer)

Dialer for small and medium organisation – Call center Dialer

HOW TO LIMIT THE INBOUND CALLS from asterisk

How to write an AGI Script – script writting for asterisk

Lifetime free open source dialer for inbound & Outbound