asterCC released version 0.21 beta

asterCRM 0.075 in asterCC 0.21 beta:

added customer_lead and note_leadadded defined code for note

added user online statistic and user online report

added user type and user privileges

added attend transfer for agent modeadded hangupcause, hangupcausetxt and dialstatus to cdr tablefixed bug of astercc daemon works with asterisk 1.2fixed bug of incorrect didnumber on agent modefieed bug of incorrect curcdr works with astercrm.agiadded doubleCheckCampaign in astercc.confadded export data to daillist directlyadded count and time of tranfer to cdr and dialedlist

asterBilling 0.15in asterCC 0.21 beta:

added pushcall mode(works with pushcall.agi)added billing by accountcode modeadded query and hangup current channels in system page

Related posts:

asterCC 0.14-beta and asterCC BOX 0.14-beta-rc1 releasedasterCC 0.14-beta and asterCC BOX 0.14-beta-rc1 releasedasterCC and asterCC-BOX released version 0.2asterCC and asterCC-BOX released version 0.2astercc 0.2 beta released

tutorial: use astercc,freepbx and asterisk to build a broadcasting system with IVR and Queue agents

the most feature in astercc is the predictive dialer, using the dialer you could improve the work efficiency. In this tutorial, I will introduce how to setup a dialer with a pre-configed IVR:  when dialer start work, customers will hear a IVR which you configed in Freepbx, so we could also config in the IVR to accept customer input, and we can forward to an new IVR or agents in a queue.

1. config a Queue in freepbx

usually you want to config a queue to resonpse customer if they want to reach some live agent, so we config a queue first

2. config an IVR in freepbx

2.1 first we will add some voice in the IVR, you’d like to use a recording software, just notice that in asterisk, it requires to use wav format and 16bit, 8000HZ mono

2.2 add a recording

you can either upload the recording you finished in your pc, or use a ext. in system to record a new one

2.3 add IVR

now queue and recording is ready, we could add the IVR

in announcement, we select the recording we just done, and also we added two options to accept customer input, when customer hits 1, he will go queue 900, and when hits 2, he will go to ext. 5001.

4. add misc

then we need add a misc so the dialer could reach the IVR

remember this feature code 800, we will use it when configing campaign in astercc, you could use some other code as you like, just make sure it’s unique in your freepbx

now we finished the job in freepbx, you’d like to dial 800 from any extension, it supposes to bring you to the IVR.

5. config the campaign in astercc

login as admin, and go to campaign, add a campaign as following

make sure in “Inexten” you put the same code as in “misc application”, and in “Queue number” put the same queue number as your freepbx queue

6. add agent user in astercrm

you’d like to add some account for your agents if you want them get a popup form when they start answer customer calls, go to “Extension” to add astercrm account for your agents, make sure the account “extension” match the ext. in your freepbx

7. start dialer and test agent

before start dialer, you’d like to check your astercc.conf

make sure this parameter is configed as

doublecheckcampaign = yes

then restart astercc daemons

/opt/asterisk/scripts/astercc/asterccd restart

ask your agent login and as admin go to dialer page, start the dialer

as an agent, he will get a popup when customer answered the call and hit 1 for queue

8. check report

you could go to dialedlist to check the campaign status

* to get a working freepbx and asterisk easily, you could choose  astercc box, elastix, pbxinaflash or trixbox

Related posts:

dialer, queue and popup for asterisk callcenter(freepbx,trixbox,elastix,pbxinflash) with asterCRMdialer, queue and popup for asterisk callcenter(freepbx,trixbox,elastix,pbxinflash) with asterCRMtutorial: use astercrm & asterisk for broadcastingtutorial: use astercrm & asterisk for broadcastingbuilding a virtual office using astercrm ,freepbx and asterisk

Login to asterisk queue as dynamic agent in astercrm agent interface

In astercrm version 0.7 ,  we can login to queue which assigned to campaigns as dynamic agent in portal page.

Set “queue number” and “queue context” for campaign:

queue context: which context will be used when login to queue, for example:exteinsion is 8000,set queue context to from-queue,then login to queue as Local/8000@from-queue/n ?

login to astercrm,the queues will display in “Campaign Pannel(Queue)”:

click login button after campaign-1(1000),will login to queue 1000 as Local/8000@from-queue/n, login button turn to logoff:

execute queue show 1000 in asterisk Cli:

then could use pause/continue to control agent if receive incoming call from queue:

Related posts:

cant login queue via agent interfaceDynamic Agent mode with asterisk, FreePBX and asterCRMdialer, queue and popup for asterisk callcenter(freepbx,trixbox,elastix,pbxinflash) with asterCRMbuilding a virtual office using astercrm ,freepbx and asterisktutorial: use astercrm & asterisk for broadcasting

Installing vzdump on CentOS 5.4 (32-bit)

There are a few items required for installing vzdump for OpenVZ on 32bit CentOS.

First, make sure you read the installation post covering the 64bit install. Here are the commands for 32bit. Here is how you get vzdump on a clean version of 32bit CentOS (via the hostnode):

rpm -ivh "ftp://ftp.pbone.net/mirror/ftp.freshrpms.net/pub/freshrpms/pub/dag/redhat/el5/en/i386/RPMS.dag/cstream-2.7.4-3.el5.rf.i386.rpm"wget http://dag.wieers.com/rpm/packages/perl-LockFile-Simple/perl-LockFile-Simple-0.206-1.el5.rf.noarch.rpmrpm -ivh perl-LockFile-Simple-0.206-1.el5.rf.noarch.rpm/bin/rm perl-LockFile-Simple-0.206-1.el5.rf.noarch.rpmrpm -ivh "http://chrisschuld.com/centos54/vzdump-1.2-6.noarch.rpm"

Make sure you add in the PERL5LIB in your export (see the 64bit post):

export PERL5LIB=/usr/share/perl5/

Asterisk 1.8.2.2 Now Available (Security Release)

Asterisk 1.8.2.2 Now Available (Security Release)

Thu, 20 Jan 2011 21:35:00 -0400 The Asterisk Development Team has announced a release for the security issue described in AST-2011-001.

Due to a failed merge, Asterisk 1.8.2.1 which should have included the security fix did not. Asterisk 1.8.2.2 contains the the changes which should have been included in Asterisk 1.8.2.1.

This releases is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2, 1.8.1.2, and 1.8.2.2 resolve an issue when forming an outgoing SIP request while in pedantic mode, which can cause a stack buffer to be made to overflow if supplied with carefully crafted caller ID information. The issue and resolution are described in the AST-2011-001 security advisory.

For more information about the details of this vulnerability, please read the security advisory AST-2011-001, which was released at the same time as this announcement.

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.2

Security advisory AST-2011-001 is available at:

http://downloads.asterisk.org/pub/security/AST-2011-001.pdf

Thank you for your continued support of Asterisk!

Asterisk 1.10 Update

Asterisk 1.10 Update
Greetings,

Shortly after the release of Asterisk 1.8, we had a developer meeting and discussed some of the projects that people would like to see in Asterisk 1.10. We discussed the schedule there a bit, as well. Now that Asterisk 1.8 has settled down and we are well into the development cycle for Asterisk 1.10, it is a good time to revisit the plans for the next release.

At Digium, the biggest thing we have been working on for 1.10 so far is replacing the media infrastructure in Asterisk. Most of the critical and invasive plumbing work is done and has been merged into trunk. Next we're looking at building up some features on top of that, such as adding more codecs, enhancing ConfBridge() to support additional sampling rates (HD conferencing), adding features that exist in MeetMe() but not Co"> .msg_list {margin: 0px;padding: 0px;width: 100%; } .msg_head {padding: 5px 10px;cursor: pointer;position: relative;background-color:#f4f4ff;border: 1px solid #ccc;margin:0px; } .msg_body {padding: 5px 10px 15px;background-color:#F4F4F8; display: none;border: 1px solid #ccc;border-top: 0px; } .msg_body_2 {padding: 5px 10px 15px;background-color:#F4F4F8; }
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Asterisk 1.10 Update

Wed, 16 Feb 2011 18:23:18 -0400 Russell Bryant has posted some information about Asterisk 1.10:

Greetings,

Shortly after the release of Asterisk 1.8, we had a developer meeting and discussed some of the projects that people would like to see in Asterisk 1.10. We discussed the schedule there a bit, as well. Now that Asterisk 1.8 has settled down and we are well into the development cycle for Asterisk 1.10, it is a good time to revisit the plans for the next release.

At Digium, the biggest thing we have been working on for 1.10 so far is replacing the media infrastructure in Asterisk. Most of the critical and invasive plumbing work is done and has been merged into trunk. Next we're looking at building up some features on top of that, such as adding more codecs, enhancing ConfBridge() to support additional sampling rates (HD conferencing), adding features that exist in MeetMe() but not ConfBridge(), and enhancing codec negotiation.

Of course, many others have been working on new developments as well. I would encourage you to respond if you'd like to provide an update on some new things that you're working on.

We would like to release Asterisk 1.10 roughly a year after Asterisk 1.8. This will be a standard release, not LTS. To have the release out in the October time frame, we need to branch off 1.10 (feature freeze) at the end of June. At that point we will begin the beta and RC process. If you're working on new development projects that you would like to get into Asterisk 1.10, please keep this timeline in mind.

As always, comments and questions are welcome.

Thanks,

--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Asterisk: The Definitive Guide book available for pre order and public review
October 21, 2010 Average Vote: 10
The 3rd edition of the Asterisk book is on its way and available for public review and pre-order.

Article number 2500
January 25, 2011 Average Vote: 10
This article marks number 2500. We have been producing articles on Asterisk and related products now since 2004.

Submit your own stories
September 21, 2010 Average Vote: 10
We have made a pretty big update to the Daily Asterisk News. You can now submit your own stories! There is now a link at the top of the Daily Asterisk News. Just click the link, fill out the form and we will check over your story and submit it.

Using locked PAP2 and PAP2-NA with Asterisk
August 23, 2005 Average Vote: 10
VoIPHacker has posted details on how to unlock a PAP2 and PAP2-NA.

Asterisk 1.10 Update
February 17, 2011 Average Vote: 10
Russell Bryant has posted some information about Asterisk 1.10.

Asterisk-Fax
May 19, 2005 Average Vote: 10
Just thought I'd provide a pointer to the Asterisk-Fax site (it came up on Asterisk-Users).

Tropo Now Speaks Asterisk Gateway Interface (AGI)
October 4, 2010 Average Vote: 10
Jason Goeke has posted a link to an article about Tropo support for Asterisk via AGI.

The Everything Asterisk Video Collection
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Steven Sokol has posted a blog entry on Asterisk Video Resources.

AstriDevCon: October 29th, Washington DC
August 23, 2010 Average Vote: 10
John Todd has posted a note about the AstriDevCon conference which occurs within the Astricon conference.

Asterisk and Kamailio (openser) realtime integration
August 5, 2010 Average Vote: 9.9
Daniel-Constantin Mierla posted a writeup on combining Asterisk and Kamailio.

Interview with Mark Spencer
November 26, 2004 Average Vote: 9.9
We have managed to get an interview with Mark Spencer AKA Markster. Mark Spencer is the creator of Asterisk and by far the most active developer.

ZORG new C++ and Java ZRTP implementation public release
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Andrea Cristofanini from PrivateWave has sent us across a press release about a new Open Source encryption implementation.

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Olle has posted an update on IPV6 in Asterisk and a link to a blog post of his.

Proposal for T.38 transparent gateway design in Asterisk
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Kevin Fleming has posted a proposed design for a transparent T.38 gateway for Asterisk:

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Hey all - I am back online after some pretty big projects which have taken all my time. Will be updating the Asterisk news over the next few days.
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February 17, 2011
Russell Bryant has posted some information about Asterisk 1.10.

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Darrick Hartman has posted details of the latest beta images for AstLinux 0.7.6 - the embedded Asterisk distribution.

Article number 2500
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This article marks number 2500. We have been producing articles on Asterisk and related products now since 2004.

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January 24, 2011
Crystal Provisioning - The best and the easiest way to deploy, configure and manage any sip phones in an Asterisk environment

People Tech CRM
January 24, 2011
PeopleTech have sent details of their CRM package.

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January 21, 2011
The Asterisk Development Team has announced a release for the security issue described in AST-2011-001.

Safi Communications Suite 1.5.0 Beta
January 20, 2011
Safi Systems are happy to announce the 1.5.0 Beta release of SafiWorkshop and SafiServer, now collectively referred to as the Safi Communications Suite (or SCS for short).

AST-2011-001: Stack buffer overflow in SIP channel driver
January 20, 2011
When forming an outgoing SIP request while in pedantic mode, a stack buffer can be made to overflow if supplied with carefully crafted caller ID information.

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Bryan M. Johns has posted details about a new weekly Asterisk developer call.

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Asterisk IAX clients

AG-168V IAXy by Digium S100-FX by X100P Virbiage 3010: Available direct from Virbiage, or rebranded (e.g. this Freshtel ATA)

American Distributor for ATCOM AG168V Gateway: http://www.iareaphone.com/ Australian ATCOM distributor: http://www.ob-wan.com/voip European reseller of ATCOM AT-320 and ATCOM AG-168V: Upnet Taide European reseller: for all ATCOM products. IP Phones, ATA's and USB phones German / European reseller: http://www.sipmax.de for all ATCOM products. AT 320 PD - SIP/IAX-Phone, AU 100 - USB-IAX-Phone, AG168V - SIP/IAX-ATA Libraries Corraleta Technology SDK Programmable web-based IAX softphone (DHTML/Java/JavaScript) IAXClient: IAX2 softphone library IaxClient SDK for building Cross Browser, Click2Call, WebPhones and SoftPhones (free for non-commercial) Java IAX Implementation, Corraleta Technology SDK GPL'ed version Pure Java IAX Implementation, GPL'ed Corraleta SDK http://callino.cc/jiaxcapplet/: Java Wrapper for the IAXClient library, including applet which does provide javascript support (Click2Dial) libiax2 : Digium's base IAX2 protocol library Public sector training for building Cross Browser, Click2Call, WebPhones and SoftPhones (free for non-commercial) Njiax IAX2 Java library IAX2 library developed in Java by Nomasystems. GNU Lesser Public License. "100% pure Java". A good example on how using this library can be found in this click to talk (with DTMF support): http://www.nomasystems.com/contacto.yaws SimpleIAX for Delphi is a component library for Delphi and C++Builder, allowing to build the VoIP applications. SimpleIAX is built on top of the IAXClient. Applications (see also VOIP Phones ) http://www.messagingbay.com/ IAX2 client for Android/BlackBerry/Web browsers. Search "MPhoneGG" on Android Market or email support@messagingbay.com for more info12/02/2011. Coccinella Multi Platform XMPP client with whiteboard. Since Version 0.96.19 with enabled IAX2 phone integrated. Corraleta Connect Crossplatform click to talk via IAX (Not free) Cubix for Windows (Available for OEM licensing) DIAX for Windows (Freeware for personal use) Ephone for Windows (May be used Commercially free of charge) GnoPhone for Linux IAX Phone for Windows IAX2 Click2Call Cross Browser, Click2Call (free for non-commercial) (Windows) IAX2 WebPhone build with IAXClientSDK for IE. (Windows) IAXComm for Windows, Linux & MacOSX iaxLite for Windows IAXOCX -a free IAX ActiveX Control for Windows. JackenIAX for MacOSX JIAXclient - JAVA applet and library Kiax for Linux and Windows LoudHush for MacOSX OctoPhone -a free IAX softphone for Windows. SFLphone Open source (GPLv3) IAX2/SIP compatible softphone for Linux Voix Manager for Windows Linux and Mac OSX, a switchboard application with an IAX/IAX2 phone integrated Voix Phone Free IAX/IAX2 Softphone for Windows, Linux & MacOSX Yate for Windows, Linux and BSD - an IAX client and server (license GPL). Zoiper - formerly Idefisk for Windows, Linux and Mac
;This text is for the iax.conf file
; Inter-Asterisk eXchange driver definition
;Register commands to connect to another IAX server has to
;be in the [general] section of iax.conf.
;To set up IAX in trunk mode, add "trunk=yes" to your definition in iax.conf.
;IAX Trunking needs support of a hardware timer. See Asterisk timer for more information.
;Do not use both "host=dynamic" and "defaultip=111.222.333.444",
; make sure it is just one or the other (see bugs 558 and 673).
;
; General settings, like port number to bind to, and
; an option address (the default is to bind to all
; local addresses).
;
[general]
port=5036
iaxcompat=yes
disallow=all
;
; All of the following codecs are supported by VoicePulse Connect!
;
allow=ulaw
allow=ilbc
allow=gsm
;allow=adpcm
;allow=alaw
;allow=g723.1
;allow=g729
jitterbuffer=no
[voicepulse]
type=friend
context=dialout
username=
secret=
host= gwiaxt01.voicepulse.com

[IareaNet.Net]
type=friend
auth=md5
username=
secret=
context=dialout
host=208.111.0.45
qualify=yes
jitterbuffer=no

[anotherprovider]
type=friend
auth=md5
username=
secret=
context=dialout
host=IAX2.anotherprovider.com
qualify=yes
jitterbuffer=yes

[8101]test
type=friend
username=8101
secret=
context=dialout
host=dynamic
callerid=Joe Useri<8101> Asterisk phones
VOIP Phones Go back to IAX