36 Great free Asterisk applications

Hi, I was looking round on the Internet and saw there was no definitive list of free applications available for use with Asterisk, so I thought I'd compile a list for you all. If there's anything that you know of that is actively maintained but not in the list below, let me know (bear in mind I'm not including distros or Asterisk packagings in this list).

Hopefully there are a few programs in the list that even the most seasoned Asterisk professionals have not seen before.

Flash Operator Panel

Flash Operator Panel is a switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your PBX activity in real time. The layout is configurable (button sizes and colors, icons, etc). You can have more than 100 buttons active per screen. On the Live Demo there are 28 buttons defined. It also supports contexts: you can have one server running and many different client displays (for hosted PBX, different departments, etc). It can integrate with CRM software, by poping up a web page (and passing the CLID) when a specified button is ringing.

PHP AGI

AGI (Asterisk Gateway Interface) is a way of running programs or scripts which are external to Asterisk. PHP AGI is a library for PHP which simplifies writing AGI scripts.

Web Meetme

Web-MeetMe is a suite of PHP pages to allow for scheduling and managing conferences on an Asterisk PBX.

Oreka

Oreka is a modular and cross-platform system for recording and retrieval of audio streams. The project currently supports VoIP and sound device based capture. Recordings metadata can be stored in any mainstream database. Retrieval of captured sessions is web based.

FreePBX
FreePBX is a full-featured PBX web application. If you've looked into Asterisk, you know that it doesn't come with any "built in" programming. You can't plug a phone into it and make it work without editing configuration files, writing dialplans, and various messing about.

FreePBX simplifies this by giving you pre-programmed functionality accessible by a user-friendly web interfaces that allows you to have a fully functional PBX pretty much straight away with no programming required.

FreePBX also comes as a distro

Areski CDR Stats

Asterisk-Stat is a visualisation layer for Asterisk CDR statistics which are pulled from a database. It provides graphs as well as allowing you to get more information on individual calls.





sipsak

sipsak is a small command line tool for developers and administrators of Session Initiation Protocol (SIP) applications. It can be used for some simple tests on SIP applications and devices.

Asterisk PhoneBook

A common shared phone book directory based on CMS/LAMP and build for Asterisk PBX, store name and number into MySQL which will be used by each workstation browser, also by telephones with embedded XML-browser feature.

Asterisk Desktop Assistant

Asterisk Desktop Assistant is a desktop call management application for Windows PCs.  Asterisk Desktop Assistant (ADA) is Digium?s first step towards offering a comprehensive Computer Telephony Integration (CTI) suite.  It makes dialing and handling phone calls simpler and faster by adding click-to-call functionality into popular desktop applications.  It also adds call notifications directly to the Windows desktop.

A2Billing

A2Billing combined with Asterisk now gives any Telecom company a very good reason to consider the A2Billing Soft-Switch over the traditional offerings for TDM and VoIP Soft-Switches as well as wholesale and IP PBX billing, particularly when you consider the cost of A2Billing ? FREE!

AstBill

AstBill is not only a free web-based, user friendly billing interface for Asterisk and VOIP. It is also a Asterisk configuration and GUI management tool and a standardized implementation of Asterisk using REALTIME and static configuration as you please.

OSLEC

Oslec is an open source high performance line echo canceller. When used with Asterisk it works well on lines where the built-in Zaptel echo canceller fails. No tweaks like rxgain/txgain or fxotrain are required. Oslec is supplied as GPL licensed C source code and is free as in speech.

Oslec partially complies with the G168 standard and runs in real time on x86 and Blackfin platforms. Patches exist to allow any Zaptel or DAHDI compatible hardware to use Oslec. It has been successfully tested on hardware ranging from single port X100P FXO cards to dual E1 systems.

AppKonference

AppKonference is a high-performance Asterisk voice/video conferencing module. It's basically a drop in replacement for meetme - although does things a little differently and doesn't require a timing source.

OutCall

OutCALL was designed as a commercial appplication allowing Asterisk users integration with Microsoft Outlook with placing and receiving phone calls.

After over 1000 downloads as a free application, Bicom Systems Ltd. has decided to offer OutCALL in open source format in order to further stimulate development of Asterisk and related open source projects.

VMukti

VMukti is leading Asterisk/ Yate enabled p2p Video IP Communications Suite for Web / PSTN. These serverless broadband ready platform enable OS community to save 90% on capital & operating costs over proprietary software for conferencing & Call Center.

Note from editor: I've personally never managed to get this working - drop me a line if you do get it all set up and going.

Asterisk-CRM

asterCRM is a call center software for asterisk based VoIP system, also it has some CRM functions. It provides useful features such as pop-up, predictive dialer, click to call, extension status .... astercrm could work with all asterisk based system.

AsterCC

astercc is a realtime billing solution for asterisk, could work with all kinds of asterisk based system and no need do any change to your original system. astercc could be used for hosted callshop solution, pbx billing solution.

IAXModem

IAXmodem is a software modem written in C that uses an IAX channel (commonly provided by an Asterisk PBX system) instead of a traditional phone line and uses a DSP library instead of DSP hardware chipsets.

Asterisk PBX Integration Zimlet

Asterisk PBX Integration Zimlet is an Extension for Zimbra Collaboration Suite. The Zimlet does Interface with the Asterisk Manager Interface to integrate with Asterisk PBX. The main focus is dial-on-click for Phone numbers inside Contacts and Emails.

AsterFax - Asterisk Email to Fax Gateway

AsterFax provides an Email to Fax gateway for Asterisk. AsterFax lets you send an email by Fax. Enter the phone no. in the 'To' address, compose your email message and click send. You can also fax a MS-Word document or other attachment.

Asterisk Monitor

Asterisk Monitor is a HTML interface that acts a operator pannel for asterisk to display user/peer status and calls. This uses a reverse AJAX, PHP and Python to originate, transfer and hangup calls, manage queues and meetme rooms.

OpenBTS
The OpenBTS Project is an effort to construct an open-source GSM basestation using the USRP and the Asterisk VoIP PBX. Our goal is to enable a new type of hybrid GSM/VoIP cellular network for greenfield deployments in the developing world.

Asterisk .NET

The Asterisk .NET library consists of a set of C# classes that allow you to easily build applications that interact with an Asterisk PBX Server (1.0/1.2/1.4 version). Both FastAGI and Manager API supported. .NET/Mono compatible.

AGX's Asterisk Extra AddOns

Maintained version of old asterisk applications ported to 1.4 that for copyright reasons cannot be included into official Digium's releases and that for some reason the author is not keeping up-to-date. The project also require Zaptel instead of DAHDI.

AstTAPI

AstTapi, an opensource Asterisk Tapi driver for windows. This allows users of TAPI compliant applications such as Outlook and Act to dial contacts directly from the application using an Asterisk PBX Server.

Asterisk-Java

A java interface for Asterisk - allows you to write software in Java which will work with Asterisk.

IAXClient

A lightweight cross platform IP telephony client using the IAX protocol, designed for use with the Asterisk open source PBX.

Asterisk Desktop Manager

A desktop application for managing Asterisk including screen pops etc - looks quite nice but haven't tried it.

Asterisk JTAPI

Asterisk-JTAPI is a JTAPI implementation for the Asterisk software PBX system. JTAPI is a provider independent programming interface for Java to build applications for computer telephony or to add support for it. JTAPI covers a wide range of usage scenarios starting from controlling a single telephone to a whole PBX system for example in call-centers.

Asterisk-JTAPI builds on top of two other projects: Asterisk-Java, which provides a Java interface to the Asterisk manager API, and, GJTAPI, which provides a general framework for JTAPI interfaces.

Astmail

Web interface to Asterisk voicemail written in php. Includes some AJAX components such as LDAP-suggest, and user-lookup. Includes screens for forward by email and sms configuration.

Druid
Druid is an open source unified communications platform, built around technology such as Asterisk, IMAP, XMPP. Druid gives your organization access to the best available IP communications platform that bringing together voicemail, VOIP, mobile phone, faxes and instant messaging.

ViciDial

VICIDIAL is a set of programs that are designed to interact with the Asterisk Open-Source PBX Phone system to act as a complete inbound/outbound call center suite.

The agent interface is an interactive set of web pages that work through a web browser to give real-time information and functionality with nothing more than an internet browser on the client computer.

Asterisk Queue/CDR Log Analyzer

The Asterisk Queue (and CDR) Log Analyzer is a set of PHP scripts which allow selecting, listing and graphing of records from the Asterisk Queue and CDR logs via a WEB interface.

For easier access to select specific log records, the Queue and CDR logs need to be in a MySQL database. Asterisk itself records (specified in a conf file) CDR data into a MySQL database table. A Python utility program called loadq.py is provided with this package which can be used to load queue log records (as they are created) into a MySQL database table.

Asterisk WEB/PHP Event Monitor

The Asterisk Event Monitor WEB/PHP Interface was created to view the current state of Asterisk and all Asterisk Events via a WEB interface. It does not poll Asterisk for these events, instead it collects them in a MySql database via an Asterisk Manager API python script. AJAX (Javascript) is used to display the events from the database almost as they occur. All code is released under the GNU GPL license.


Crystal Recording Interface

CRI (Crystal Clear Recording Interface) provides an intelligence web interface to track is recordings and his voice mail

Features :

Access your voicemail recordings.
Setup your voice mail box
View summary of your incoming outgoing calls
Search calls and recordings by day and time.
Call monitor recordings.

Xivo

XiVO is a full PBX solution based on Asterisk with a user-friendly web interface, provisioning tools for many types of phones, CTI daemon and CTI client for Windows, Linux and MacOS. All solutions are released under GPLv3. Currently only released in French, but easy to translate in other languages, XiVO provides administrators with a simple to configure phone system.

You can find a demo on https://demo.xivo.fr with login root and password proformatique so you can try it out for yourself.

On the first page you have a dashboard for monitoring all processes and you can choose PBX on the menu to administer your PBX. It's not based on Freepbx or other existing software, it's a full development from scratch.

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Asterisk Commands throught PHP script

<pre><table width="100%" style="border: 2px solid #CCCCCC;" cellpadding="3" cellspacing="3">
<tr bgcolor="#666666">
    <td align="center" background="stripe.png"><font color="#FFFFFF"><b><h1><?php system("hostname"); ?></h1></b></font></td>
  </tr>
  <tr bgcolor="#000000">
    <td><font color="#FFFFFF"><b>Uptime:</b></font></td>
  </tr>
  <tr bgcolor="#FFE4E1">
    <td><?php system("uptime"); ?></td>
  </tr>
  <tr bgcolor="#000000">
    <td><font color="#FFFFFF"><b>System Information:</b></font></td>
  </tr>
  <tr bgcolor="#FFDAB9">
    <td><?php system("uname -a"); ?></td>
  </tr>
 <tr bgcolor="#000000">
    <td><font color="#FFFFFF"><b>Memory Usage (MB):</b></font></td>
  </tr>
  <tr bgcolor="#FFF0F5">
    <td><?php system("free -m"); ?></td>
  </tr>
 <tr bgcolor="#000000">
    <td><font color="#FFFFFF"><b>Disk Usage:</b></font></td>
  </tr>
  <tr bgcolor="#E6E6FA">
    <td><?php system("df -h"); ?></td>
  </tr>
 <tr bgcolor="#000000">
    <td><font color="#FFFFFF"><b>CPU Information:</b></font></td>
  </tr>
  <tr bgcolor="#F0FFF0">
    <td><?php system("cat /proc/cpuinfo | grep \"model name\\|processor\""); ?></td>
  </tr>
</table>

</pre>

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Configuration file for Asterisk SIP channels, for both inbound and outbound calls.

Asterisk as a SIP client
;***********For INBOUND Only*********************
register => 1001:password@mysipprovider.com/1234

;***********FOR INBOUND as well as OUTBOUND*********************
[SIPTRUNK]
fromuser=username or ip from where to send calls
host = remote-ip-address (where to send calls)
username=username
secret=any-passsword
fromdomain=remote-ip-address (Valid only when in [general] section or
type=peer.)
disallow=all
allow=g729
type = friend (user, peer, friend)
dtmfmode=rfc2833
dtmf=rfc2833
qualify = yes
rfc2833compensate=yes
insecure=port,invite
qualify = yes
canreinvite=no
restrictcid=yes
context = sip-call-in ( this section will be defined in
extensions.conf and If type=user, the Context for the inbound call
from this SIP user definition. If type=peer, the Context in the
dialplan for outbound calls from this SIP peer definition. If
type=friend the context used for both inbound and outbound calls
through the SIP entities definition. If no type=user entry matches an
inbound call, then a type=peer or type=friend will match if the
hostname or IP address defined in host= matches.)
In extensions.conf you'd then use a statement like this:.
;***********For OUTBOUND*********************
exten => _X.,1,Set(CDR(accountcode)=SIPTRUNK100)
exten => _X.,n,Dial(SIP/${EXTEN}@SIPTRUNK,60)
exten => h,n,Hangup()
;***********FOR INBOUND as well as OUTBOUND*********************
[sip-call-in]
exten => 1234,1,Answer ; 1234 is the contact extension, default
contact extension is "s"
exten => 1234,2,Dial(SIP/111,25,Ttr) ; incoming calls are redirected
to SIP telephone with number 111
exten => 1234,3,Hangup

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Asterisk Detailed Variable List

Asterisk standard channel variables

There are a number of variables that are defined or read

by Asterisk. Here is a list of them. More information is

available in each application's help text. All these variables

are in UPPER CASE only.

to diaplay all channel variable use this command

DumpChan(<min_verbose_level>)

Variables marked with a * are builtin functions and can't be set,

only read in the dialplan. Writes to such variables are silently

ignored.

${ACCOUNTCODE} * Account code (if specified) (Deprecated; use ${CDR(accountcode)})

${BLINDTRANSFER} The name of the channel on the other side of a blind transfer

${BRIDGEPEER} Bridged peer

${CALLERANI} * Caller ANI (PRI channels) (Deprecated; use ${CALLERID(ani)})

${CALLERID} * Caller ID (Deprecated; use ${CALLERID(all)})

${CALLERIDNAME} * Caller ID Name only (Deprecated; use ${CALLERID(name)})

${CALLERIDNUM} * Caller ID Number only (Deprecated; use ${CALLERID(num)})

${CALLINGANI2} * Caller ANI2 (PRI channels)

${CALLINGPRES} * Caller ID presentation for incoming calls (PRI channels)

${CALLINGTNS} * Transit Network Selector (PRI channels)

${CALLINGTON} * Caller Type of Number (PRI channels)

${CHANNEL} * Current channel name

${CONTEXT} * Current context

${DATETIME} * Current date time in the format: DDMMYYYY-HH:MM:SS (Deprecated; use ${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)})

${DB_RESULT} Result value of DB_EXISTS() dial plan function

${DNID} * Dialed Number Identifier (Deprecated; use ${CALLERID(dnid)})

${EPOCH} * Current unix style epoch

${EXTEN} * Current extension

${ENV(VAR)} Environmental variable VAR

${GOTO_ON_BLINDXFR} Transfer to the specified context/extension/priority

after a blind transfer (use ^ characters in place of

| to separate context/extension/priority when setting

this variable from the dialplan)

${HANGUPCAUSE} * Asterisk cause of hangup (inbound/outbound)

${HINT} * Channel hints for this extension

${HINTNAME} * Suggested Caller*ID name for this extension

${INVALID_EXTEN} The invalid called extension (used in the "i" extension)

${LANGUAGE} * Current language (Deprecated; use ${LANGUAGE()})

${LEN(VAR)} * String length of VAR (integer)

${PRIORITY} * Current priority in the dialplan

${PRIREDIRECTREASON} Reason for redirect on PRI, if a call was directed

${RDNIS} * Redirected Dial Number ID Service (Deprecated; use ${CALLERID(rdnis)})

${TIMESTAMP} * Current date time in the format: YYYYMMDD-HHMMSS (Deprecated; use ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})

${TRANSFER_CONTEXT} Context for transferred calls

${FORWARD_CONTEXT} Context for forwarded calls

${UNIQUEID} * Current call unique identifier

${SYSTEMNAME} * value of the systemname option of asterisk.conf

Application return values

In Asterisk 1.2, many applications return the result in a variable

instead of, as in Asterisk 1.0, changing the dial plan priority (+101).

For the various status values, see each application's help text.

${AGISTATUS} * agi()

${AQMSTATUS} * addqueuemember()

${AVAILSTATUS} * chanisavail()

${CHECKGROUPSTATUS} * checkgroup()

${CHECKMD5STATUS} * checkmd5()

${CPLAYBACKSTATUS} * controlplayback()

${DIALSTATUS} * dial() - see also ${HANGUPCAUSE}

${DBGETSTATUS} * dbget()

${ENUMSTATUS} * enumlookup()

${HASVMSTATUS} * hasnewvoicemail()

${LOOKUPBLSTATUS} * lookupblacklist()

${OSPAUTHSTATUS} * ospauth()

${OSPLOOKUPSTATUS} * osplookup()

${OSPNEXTSTATUS} * ospnext()

${OSPFINISHSTATUS} * ospfinish()

${PARKEDAT} * parkandannounce()

${PLAYBACKSTATUS} * playback()

${PQMSTATUS} * pausequeuemember()

${PRIVACYMGRSTATUS} * privacymanager()

${QUEUESTATUS} * queue()

${RQMSTATUS} * removequeuemember()

${SENDIMAGESTATUS} * sendimage()

${SENDTEXTSTATUS} * sendtext()

${SENDURLSTATUS} * sendurl()

${SYSTEMSTATUS} * system()

${TRANSFERSTATUS} * transfer()

${TXTCIDNAMESTATUS} * txtcidname()

${UPQMSTATUS} * unpausequeuemember()

${VMSTATUS} * voicmail()

${VMBOXEXISTSSTATUS} * vmboxexists()

${WAITSTATUS} * waitforsilence()

Various application variables

${CURL} * Resulting page content for curl()

${ENUM} * Result of application EnumLookup

${EXITCONTEXT} Context to exit to in IVR menu (app background())

or in the RetryDial() application

${MONITOR} * Set to "TRUE" if the channel is/has been monitored (app monitor())

${MONITOR_EXEC} Application to execute after monitoring a call

${MONITOR_EXEC_ARGS} Arguments to application

${MONITOR_FILENAME} File for monitoring (recording) calls in queue

${QUEUE_PRIO} Queue priority

${QUEUE_MAX_PENALTY} Maximum member penalty allowed to answer caller

${QUEUESTATUS} Status of the call, one of:

(TIMEOUT | FULL | JOINEMPTY | LEAVEEMPTY | JOINUNAVAIL | LEAVEUNAVAIL)

${RECORDED_FILE} * Recorded file in record()

${TALK_DETECTED} * Result from talkdetect()

${TOUCH_MONITOR} The filename base to use with Touch Monitor (auto record)

${TOUCH_MONITOR_FORMAT} The audio format to use with Touch Monitor (auto record)

${TOUCH_MONITOR_OUTPUT} * Recorded file from Touch Monitor (auto record)

${TXTCIDNAME} * Result of application TXTCIDName

${VPB_GETDTMF} chan_vpb

The MeetMe Conference Bridge uses the following variables:

${MEETME_RECORDINGFILE} Name of file for recording a conference with

the "r" option

${MEETME_RECORDINGFORMAT} Format of file to be recorded

${MEETME_EXIT_CONTEXT} Context for exit out of meetme meeting

${MEETME_AGI_BACKGROUND} AGI script for Meetme (zap only)

${MEETMESECS} * Number of seconds a user participated in a MeetMe conference

The VoiceMail() application uses the following variables:

${VM_CATEGORY} * Sets voicemail category

${VM_NAME} * Full name in voicemail

${VM_MAILBOX} * Mailbox Number

${VM_DUR} * Voicemail duration

${VM_MSGNUM} * Number of voicemail message in mailbox

${VM_CALLERID} * Voicemail Caller ID (Person leaving vm)

${VM_CIDNAME} * Voicemail Caller ID Name

${VM_CIDNUM} * Voicemail Caller ID Number

${VM_DATE} * Voicemail Date

${VM_MESSAGEFILE} * Path to message left by caller

The VMAuthenticate() application uses the following variables:

${AUTH_MAILBOX} * Authenticated mailbox

${AUTH_CONTEXT} * Authenticated mailbox context

DUNDiLookup() uses the following variables

${DUNDTECH} * The Technology of the result from a call to DUNDiLookup()

${DUNDDEST} * The Destination of the result from a call to DUNDiLookup()

The Zaptel channel sets the following variables:

${ANI2} * The ANI2 Code provided by the network on the incoming call. (ie, Code 29 identifies call as a Prison/Inmate Call) See also: NANPA ANI II Digits Assignments

${CALLTYPE} * Type of call (Speech, Digital, etc)

${CALLEDTON} * Dialplan for called number on PRI/BRI calls (17=international, 33=national, 65=local, 73=private, 0=unknown). Note: this is a misnomer, TON != dialplan.

${CALLINGSUBADDR} * Called PRI Subaddress

${FAXEXTEN} * The extension called before being redirected to "fax"

${PRIREDIRECTREASON} * Reason for redirect, if a call was directed

${SMDI_VM_TYPE} * When an call is received with an SMDI message, the 'type'

of message 'b' or 'u'

The SIP channel uses the following variables:

${SIPCALLID} * SIP Call-ID: header verbatim (for logging or CDR matching)

${SIPDOMAIN} * SIP destination domain of an inbound call (if appropriate)

${SIPUSERAGENT} * SIP user agent (THIS DOESNT WORK IN 1.8(+?), I DONT KNOW IF ANY OF THESE WORK EXCEPT FOR SIPURI)

${SIPURI} * SIP uri

${SIP_CODEC} Set the SIP codec for a call

${SIP_URI_OPTIONS} * additional options to add to the URI for an outgoing call

${RTPAUDIOQOS} RTCP QoS report for the audio of this call

${RTPVIDEOQOS} RTCP QoS report for the video of this call

The Agent channel uses the following variables:

${AGENTMAXLOGINTRIES} Set the maximum number of failed logins

${AGENTUPDATECDR} Whether to update the CDR record with Agent channel data

${AGENTGOODBYE} Sound file to use for "Good Bye" when agent logs out

${AGENTACKCALL} Whether the agent should acknowledge the incoming call

${AGENTAUTOLOGOFF} Auto logging off for an agent

${AGENTWRAPUPTIME} Setting the time for wrapup between incoming calls

${AGENTNUMBER} * Agent number (username) set at login

${AGENTSTATUS} * Status of login ( fail | on | off )

${AGENTEXTEN} * Extension for logged in agent

The Dial() application uses the following variables:

${DIALEDPEERNAME} * Dialed peer name

${DIALEDPEERNUMBER} * Dialed peer number

${DIALEDTIME} * Time for the call (seconds)

${ANSWEREDTIME} * Time from dial to answer (seconds)

${DIALSTATUS} * Status of the call, one of:

(CHANUNAVAIL | CONGESTION | BUSY | NOANSWER

| ANSWER | CANCEL | DONTCALL | TORTURE)

${DYNAMIC_FEATURES} * The list of features (from the applicationmap section of

features.conf) to activate during the call, with feature

names separated by '#' characters

${LIMIT_PLAYAUDIO_CALLER} Soundfile for call limits

${LIMIT_PLAYAUDIO_CALLEE} Soundfile for call limits

${LIMIT_WARNING_FILE} Soundfile for call limits

${LIMIT_TIMEOUT_FILE} Soundfile for call limits

${LIMIT_CONNECT_FILE} Soundfile for call limits

${OUTBOUND_GROUP} Default groups for peer channels (as in SetGroup)

    * See "show application dial" for more information

The chanisavail() application sets the following variables:

${AVAILCHAN} * the name of the available channel if one was found

${AVAILORIGCHAN} * the canonical channel name that was used to create the channel

${AVAILSTATUS} * Status of requested channel

When using macros in the dialplan, these variables are available

${MACRO_EXTEN} * The calling extensions

${MACRO_CONTEXT} * The calling context

${MACRO_PRIORITY} * The calling priority

${MACRO_OFFSET} Offset to add to priority at return from macro

The ChanSpy() application uses the following variables:

${SPYGROUP} * A ':' (colon) separated list of group names.

(To be set on spied on channel and matched against the g(grp) option)

If you compile with OSP support, these variables are used:

${OSPINHANDLE} OSP handle of in_bound call

${OSPINTIMELIMIT} Duration limit for in_bound call

${OSPOUTHANDLE} OSP handle of out_bound call

${OSPTECH} OSP technology

${OSPDEST} OSP destination

${OSPCALLING} OSP calling number

${OSPOUTTOKEN} OSP token to use for out_bound call

${OSPOUTTIMELIMIT} Duration limit for out_bound call

${OSPRESULTS} Number of remained destinations

Call File extension variables:

${REASON} The reason why an auto-dialout call failed

CDR Variables

If the channel has a cdr, that cdr record has it's own set of variables which

can be accessed just like channel variables. The following builtin variables

are available and, unless specified, read-only.

${CDR(clid)} Caller ID

${CDR(src)} Source

${CDR(dst)} Destination

${CDR(dcontext)} Destination context

${CDR(channel)} Channel name

${CDR(dstchannel)} Destination channel

${CDR(lastapp)} Last app executed

${CDR(lastdata)} Last app's arguments

${CDR(start)} Time the call started.

${CDR(answer)} Time the call was answered.

${CDR(end)} Time the call ended.

${CDR(duration)} Duration of the call.

${CDR(billsec)} Duration of the call once it was answered.

${CDR(disposition)} ANSWERED, NO ANSWER, BUSY

${CDR(amaflags)} DOCUMENTATION, BILL, IGNORE etc

${CDR(accountcode)} The channel's account code (read-write).

${CDR(uniqueid)} The channel's unique id.

${CDR(userfield)} The channels uses specified field (read-write).

In addition, you can set your own extra variables with a traditional

Set(CDR(var)=val) to anything you want.

NOTE Some CDR values (eg: duration & billsec) can't be accessed until the call has terminated. As of 91617, those values will be calculated on-demand if requested. Until that makes it into a stable release, you can set endbeforehexten=yes in cdr.conf, and then use the "hangup" context to wrap up your call.

Certain functional variables may be accessed with ${foo(<args>)}. A list

of these functional variables may be found by typing "show functions"

at the Asterisk CLI.

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Display Post Title Before Blog Title in Blogger for SEO

By default Blogger displays the blog title before post title. For an example <Blog Title>: <post title>.In terms of SEO this is bad. We want to display the post title first and then the blog title. To change the default behavior follow the following steps.

• Log into Blogger and go the "Design" tab
• Then click on "Edit HTML" link.
• Make sure you take a back of your existing template by clicking on "Download Full Template" link and saving the XML file to a safe storage location.
• Search for the following text in the template.

<title><data:blog.pageTitle/></title>

Then replace the above line in red with the following lines

<b:if cond='data:blog.pageType == &quot;item&quot;'> <title><data:blog.pageName/> | <data:blog.title/></title> <b:else/> <title><data:blog.pageTitle/></title> </b:if>

Finally save the change by clicking on "SAVE TEMPLATE" button and refresh your blog to see the results

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