Elastix Beta with Asterisk 1.8 Available

Elastix Beta with Asterisk 1.8 Available

As we announced at the end of April, we are launching today an update for Asterisk 1.8 in Elastix.

This update comes installed by default on 2.0.4-B4 our newest beta version of Elastix. There is an update available also for 2.0.3 version in the Elastix-Beta repositories, that can be can be activated at the Elastix Web Interface (System->Updates->Repositories). This update is available for all users of that version who want to test this new launch.

Elastix 2.0.4-B4 also includes Dahdi 2.4.1.2

Among the features you will find on Elastix are the following:

- Support for Secure RTP
- Support for IPv6
- Support for Google Talk and Google Voice
- Calendar Integration with support for CalDav, iCalendar, and others

Elastix 2.0.4-B4 - http://elx.ec/204beta4

We wait for your comments and feedback on this trial phase.

 

"After testing Elastix with OpenVox hardware, we certainly recommend this good product for PBX projects"

Belief Mo

OpenVox Communications Co.

Hong Kong

PaloSanto Solutions silver sponsor at Asterisk World

PaloSanto Solutions silver sponsor at Asterisk World

Elastix and PaloSanto Solutions will be present at the next ITExpo East on the 2nd of February, 2011.

ITExpo is part of a dual event that TMC develops in Miami and Los Angeles, at the beginning and midyear, respectively. ITExpo gathers the most representative exponents of the telecommunication industry with keynote conferences and product exhibition.

Elastix will be present and you will have the opportunity to visit our stand and talk with important members of the Project and Gold Resellers of the region. See exhibitor list.

As part of our participation we will make the launch of some products, enlist the certified hardware available at the moment and will announce the 2010 community awards winners.

Elastix is also Silver Sponsor for Asterisk World, a collocated event inside ITExpo and dedicated to this important telephony distribution. We will present a keynote there the 3rd of February at 12h30 Miami.

http://www.tmcnet.com/voip/conference/digium-asterisk-world/east-11/

Join us at this event, we will be at Booth #1009. For a free pass to visit our booth you can register here

We hope to see you all

Sincerely

 

"Elastix is a stable, reliable and flexible system, built to serve both: integrators and end users. For these reasons we chose it as the default operating system in our IP-PBX solutions, and market feedback confirms this was a good decision."

Eran Gal - CEO

Xorcom Ltd

Preparing for Asterisk on Ubuntu 10.10

Through the next few posts, I plan on showing you how to install Asterisk 1.6.x on Ubuntu 10.10, link it to an Avaya S8700/8300 through H.323 and configure it to successfully handle all voicemail.  This project is a great solution for anyone still using an Octel voicemail box, as support is quickly coming to an end (June 2011).  This solution solves a few problems others have been having – including MWI (message waiting indicators) on Avaya telephones AND figuring out what extension was dialed.

I plan on showing an exact step-by-step on my setup, so duplicating it should be very straight forward.   Let’s start with the installation.To begin, you’re going to need a copy of Ubuntu 10.10  Before we go any further, I’m sure there are plenty of users that would love to debate the usage of a certain flavor of *nix over another.  I’m not interested.  I’ve worked with Red Hat, CentOS and Mandrake/Mandriva, but *most* of my experience has been with Ubuntu.  I first installed Dapper Drake on my home machine years ago, and forced myself to become familiar with a Windows alternative.  I am NOT a Linux power user, but feel fairly comfortable with Ubuntu – so I choose to use it for my Asterisk solution.

Installing Ubuntu should be cake, and reproducing a walk-through for this would be repetitive and unnecessary.

After installing Ubuntu, we’re going to need a few packages/libraries prior to installing Asterisk.  If your install requires more than setting up Asterisk as just a voicemail server, you may need additional packages.  In some future posts, I’ll go over this along with step by step instructions on getting things to work (for example – I have my Asterisk box working with Google Voice.  This required 2 more packages and recompiling Asterisk.)

Let’s start by getting a command prompt, and elevating our permissions.

sudo su

Next, let’s install some packages (some might already be installed)

apt-get install ssh
apt-get install gcc
apt-get install ncurses-dev
apt-get install g++
apt-get install flex
apt-get install libcurl3-dev
apt-get install libxml2-dev
apt-get install libiksemel-dev
apt-get install build-essential

Rockin and rollin now!  The system is getting close for our Asterisk install.

If you plan to allow voicemail users to receive voicemail via email, we’ll also want to install Postfix.  To do that type:

apt-get install postfix

During the install of postfix, you will be asked a few questions about how you want Postfix to behave.   On my system, I set Postfix up as a relay server, forwarding mail to our Exchange boxes to be processed.

Optionally, you may want to install Postgres or MySQL to store CDR (call data records) or leverage Asterisk’s Realtime component.  I will cover this in a later post, as my setup uses MySQL to store all voicemail box information – making management a breeze and eliminating the need to reload the voicemail configuration files after each modification.

Asterisk PBX Total Cost of Ownership

Based on Asterisk, the most widely used open source telephony software, Switchvox provides all the functionality and stability of a sophisticated IP PBX platform at a fraction of the cost.

In-Stat predicts that by 2013, nearly 80 percent of businesses are expected to have adopted voice over IP (VoIP) phone systems, a rapid rise from half that number in 2009. Full-featured unified communications systems combine VoIP, messaging and presence with business data to improve employee efficiency. However, traditional UC vendors can charge SMBs big fees—in some cases, hundreds of thousand of dollars over a five-year period—for their phone systems. Most of this cost is the result of fees for system “add-ons.” This includes fees for adding separate voicemail servers, fees for mobile accessibility, fees for integration with other business software or CRM systems, and charging for critical features such as conferencing, interactive voice response (IVR) and call queues.

Leslie Conway, vice president of global marketing at Digium, commented: “When SMBs are evaluating UC solutions, it’s worthwhile to do a feature by feature comparison and evaluate the associated costs for the number of users needing those features. The potential savings on that total investment could translate to $80,000 on the average phone system over time. Our customers are in the unique position of making a buying decision that could eliminate $50,000 or more off their initial investment, while gaining enterprise-class features from a true unified communications solution.”

NP Information Systems, a Seattle-based Digium certified partner and reseller of Switchvox, often works with its SMB customers to compare features across business phone systems and identify potential cost savings. A recent customer, Clear Creek Telephone & Television, needed to replace its phone system with VoIP functionality but also required substantial customization. Formed as a co-op in the 1900s, the company has a century of experience providing telephone, cable and Internet service to businesses and residents of Oregon City and Redland, Oregon. “They’d first considered a ShoreTel system, but getting all the features they needed would have exceeded their budget,” said Vince Preece of NP Information Systems.

Asterisk 1.4.40, 1.6.2.17 and 1.8.3 released.

The Asterisk Development Team has announced the release of Asterisk 1.4.40, 1.6.2.17 and 1.8.3.

Asterisk 1.4.40

The Asterisk Development Team has announced the release of Asterisk 1.4.40. This release is available for immediate download at

http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.4.40 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* Correct issue where res_config_odbc could populate fields with invalid data.
(Closes issue #18251, #18279. Reported by bcnit, zerohalo. Tested by trev, jthurman, elguero, zerohalo. Patched by tilghman)

* Resolve issue where re-transmissions of SUBSCRIBE could break presence.
(Closes issue #18075. Reported by mdu113. Patched by twilson)

* Resolve issue in res_odbc where it may crash when a query fails.
(Closes issue #18243. Reported, patched by ks3)

* Fix CPU spike when pressing DTMF after agent login.
(Closes issue #18130. Reported by rgj. Patched by jpeeler)

* Fix cross-compiling issue.
(Closes issue #18301. Reported, patched by abelbeck)

* This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems.
(Patched by tilghman)

* Resolve several issues with DTMF based attended transfers.
(Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchaun, grecco. Patched by rmudgett).
NOTE: Be sure to read the ChangeLog for more information about these changes.

* Fix regression that changed behavior of queues when ringing a queue member.
(Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)

Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at

http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.40

Thank you for your continued support of Asterisk!

Asterisk 1.6.2.17

The Asterisk Development Team has announced the release of Asterisk 1.6.2.17. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.17 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* Resolve duplicated data in the AstDB when using DIALGROUP()
(Closes issue #18091. Reported by bunny. Patched by tilghman)

* Correct issue where res_config_odbc could populate fields with invalid data.
(Closes issue #18251, #18279. Reported by bcnit, zerohalo. Tested by trev, jthurman, elguero, zerohalo. Patched by tilghman)

* When using cdr_pgsql the billsec field was not populated correctly on unanswered calls.
(Closes issue #18406. Reported by joscas. Patched by tilghman)

* Resolve issue where re-transmissions of SUBSCRIBE could break presence.
(Closes issue #18075. Reported by mdu113. Patched by twilson)

* Fix regression causing forwarding voicemails to not work with file storage.
(Closes issue #18358. Reported by cabal95. Patched by jpeeler)

* This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems.
(Patched by tilghman)

* Resolve several issues with DTMF based attended transfers.
(Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchaun, grecco. Patched by rmudgett).
NOTE: Be sure to read the ChangeLog for more information about these changes.

* Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi.
(Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47)

* Fix regression that changed behavior of queues when ringing a queue member.
(Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)

Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at

http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.17

Thank you for your continued support of Asterisk!

Asterisk 1.8.3

The Asterisk Development Team has announced the release of Asterisk 1.8.3. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.3 resolves several issues reported by the community and would have not been possible without your participation. Thank you!

The following is a sample of the issues resolved in this release:

* Resolve duplicated data in the AstDB when using DIALGROUP()
(Closes issue #18091. Reported by bunny. Patched by tilghman)

* Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses.
(Closes issue #18464. Reported, patched by IgorG)

* Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for the function that does the parsing.
(Closes issue #18350. Reported by gbour. Patched by Marquis)

* When using cdr_pgsql the billsec field was not populated correctly on unanswered calls.
(Closes issue #18406. Reported by joscas. Patched by tilghman)

* Resolve memory leak in iCalendar and Exchange calendaring modules.
(Closes issue #18521. Reported, patched by pitel. Tested by cervajs)

* This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems.
(Patched by tilghman)

* Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi.
(Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47)

* Resolve a memory leak when the Asterisk Manager Interface is disabled.
(Reported internally by kmorgan. Patched by russellb)

* Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486.
(Reported internally. Patched by mnicholson)

* Fix regression that changed behavior of queues when ringing a queue member.
(Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)

* Resolve deadlock involving REFER.
(Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.)

Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at
http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3

Thank you for your continued support of Asterisk!

Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available

Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
These releases are available for immediate download at

http://downloads.asterisk.org/pub/telephony/asterisk/releases

The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an issue that when decoding UDPTL packets, multiple stack and heap based arrays can be made to overflow by specially crafted packets. Systems configured for T.38 pass through or termination are vulnerable. The issue and resolution are described in the AST-2011-002 security advisory.

For more information about the details of this vulnerability, please read the security advisory AST-2011-002, which was released at the same time as this announc"> .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 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available

Tue, 22 Feb 2011 17:12:23 -0400 The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4.

These releases are available for immediate download at

http://downloads.asterisk.org/pub/telephony/asterisk/releases

The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an issue that when decoding UDPTL packets, multiple stack and heap based arrays can be made to overflow by specially crafted packets. Systems configured for T.38 pass through or termination are vulnerable. The issue and resolution are described in the AST-2011-002 security advisory.

For more information about the details of this vulnerability, please read the security advisory AST-2011-002, which was released at the same time as this announcement.

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

ChangeLog-1.4.39.2
ChangeLog-1.6.1.22
ChangeLog-1.6.2.16.2
ChangeLog-1.8.2.4

Security advisory AST-2011-002 is available at:

http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

Thank you for your continued support of Asterisk!

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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.

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.

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Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
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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:

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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

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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