Latest News About Asterisk » asterisk

Asterisk 1.4.35-Rc1 and Asterisk 1.6.2.11-Rc1 Now Available

admin — Fri, 23 Jul 2010 17:45:02 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced release candidates of Asterisk for versions 1.4.35 and 1.6.2.11. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

These release candidates address issues that were reported by the community and resolved since the last round of bug fix releases.

The following is a sampling of issues resolved in these release candidates:

Send DialPlanComplete as a response, not as a separate event.
(Closes issue #17504. Reported, patched by rrb3942)
Ensure channel placed into MeetMe in ringing state is properly hung up.
(Closes issue #15871. Reported, patched by Ivan)
Fix a problem with RFC 2833 DTMF not being accepted.
(Closes issue #17571. Reported by mdeneen. Tested by richardf, maxochoa, JJCinAZ)
Add option to not do a call forward on a 482 Loop Detected.
(Reviewboard: https://reviewboard.asterisk.org/r/764/ )
Fix logging message for a stale nonce.
(Closes issue #17582. Reported, patched by kenner)
Fix some issues related to dynamic feature groups in features.conf
(Closes issue #17589. Reported,tested by lmadsen. Patched,tested by russell)
cdr_pgsql does not detect when a table is found. Add an ERROR message letting you know when a failure to get the columns from the database exists.
(Closes issue #17478. Reported, patched by kobaz)
Delete IMAP messages in reverse order to ensure reordering after each expunge does not cause deletion of the wrong message.
(Closes issue #16350. Reported by noahisaac. Patched by tilghman)

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

Issues found in any of these release candidates should be reported to the Asterisk issue tracker at https://issues.asterisk.org

Thank you for your continued support of Asterisk!

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Asterisk 1.6.2.10 Now Available

admin — Fri, 23 Jul 2010 17:11:29 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the release of Asterisk 1.6.2.10.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.10 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following are a few of the issues resolved by community developers:

Allow users to specify a port for DUNDI peers.
(Closes issue #17056. Reported, patched by klaus3000)
Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set.
(Closes issue #16815. Reported, patched by rain)
If there is realtime configuration, it does not get re-read on reload unless the config file also changes.
(Closes issue #16982. Reported, patched by dmitri)
Send AgentComplete manager event for attended transfers.
(Closes issue #16819. Reported, patched by elbriga)
Correct manager variable ‘EventList’ case.
(Closes issue #17520. Reported, patched by kobaz)

In addition, changes to res_timing_pthread that should make it more stable have also been implemented.

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10

Thank you for your continued support of Asterisk!

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Asterisk 1.4.34 Now Available

admin — Fri, 23 Jul 2010 17:10:57 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the release of Asterisk 1.4.34.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.4.34 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following are a few of the issues resolved by community developers:

Allow users to specify a port for DUNDi peers.
(Closes issue #17056. Reported, patched by klaus3000)
Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set.
(Closes issue #16815. Reported, patched by rain)
First caller into a dynamic conference new enters the pin once.
(Closes issue #15878. Reported, patched by pabelanger)
Send AgentComplete manager events in the event of blind and attended transfers.
(Closes issue #16819. Reported, patched by elbriga)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.34

Thank you for your continued support of Asterisk!

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Asterisk 1.4.34-Rc2 and Asterisk 1.6.2.10-Rc2 Now Available

admin — Wed, 14 Jul 2010 07:57:48 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the 2nd set of release candidates of Asterisk for versions 1.4.34 and 1.6.2.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

These release candidates address issues that were reported by the community and resolved since the last round of bug fix releases.

The following is are issues resolved in these release candidates:

Fix problem with RFC 2833 DTMF not being accepted.A recent check was added to ensure that we did not erroneously detect duplicate DTMF when we received packets out of order.
The problem was that the check did not account for the fact that the seqno of an RTP stream will roll over back to 0 after hitting 65535. Now, we have a secondary check that will ensure that the
seqno rolling over will not cause us to stop accepting DTMF.
(Closes issue #17571)
Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial
(Closes issue #17592)

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

Issues found in any of these release candidates should be reported to the Asterisk issue tracker at https://issues.asterisk.org

Thank you for your continued support of Asterisk!

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The IVR Clinic with Allison Smith

admin — Mon, 12 Jul 2010 12:28:22 +0000

Allison Smith, The Asterisk Voice

The 15 Commandments of IVR

Commandment #3: Keep Things Simple

So far in this Blog Series, we’ve covered the first two “commandments” of writing clean, easy-to-use IVR systems:

Commandment #1: Don’t Overestimate Your Listener’s Attention Span
Commandment #2: Thou Shalt Not Create Fake Mailboxes.

Both really key points: many writers of IVR systems feel like they have their callers “captive”, and that now that they have them listening them to their menu of options, now is a great opportunity to sell to them; to educate them in detail, and to reassure them that they’ve made the right decision by calling their company (which can be artificially made to sound bigger and more impressive than it actually may be by creating fake options and forcing callers to listen to the entire selection of options which, essentially, go nowhere.)

The customers who call into your business are busy people. They are probably over-stressed multi-taskers who simply want to accomplish what they need to accomplish in this call and move along. Your job – as the constructor of the telephone systems which “sorts” callers into appropriate departments – is to make their experience in your IVR as simplified and efficient as possible – hence Commandment #3: Keep Things Simple.

If it feels like I’m belaboring the point of simplicity, brevity, and clarity to death, I likely am. As someone who on a daily basis voices systems for a myriad of companies, I can tell you that I always have in mind – while I’m voicing the prompts – how it will feel when someone will call into this system. Will they let out a sigh, dejected, as they realize that in order to get to the department they need to speak with, they will *first* have to endure a commercial, emphasizing the benefits and wonderments of the company they’ve dialed? Will they become overwhelmed and confused by too many options – or options which are so similar as to confuse the decision of which to press? Will their selection be filtered down into too many confusing subsets?

It goes both ways: you will want to ensure that the information you’re asking for from callers is information which will not overload your organization, or make it a challenge to follow through. Just last week, I read a mailbox greeting which instructed the callers: “…For a faster response, please leave your name, number, and brief message explaining why you’re interested in partnering with us, along with your commitment level, your main passion, and the reason why you have decided to enter our industry.” You could really be inviting trouble there; most people wouldn’t likely take advantage of the situation and leave a half-hour long manifesto. But a surprisingly large number will. You need to invite that same clarity, brevity, and economy in a request for incoming information if you have any hope of boiling down the information gathered into a useful form and following through.

Reduce down the choices into the simplest options. Get callers to their needed department as quickly as possible. Don’t ask for information to be input – such as pin or account numbers – if the live agent is just going to ask for the information again. And above all: respect the caller’s time and energy.

Next blog: we’re going to be drilling deeper into the mechanics of sorting your callers into various departments, by giving them the option of not participating in the format you’ve designed at all. That’s right: we’ll be tackling Commandment #4: Always Give Callers an Opt-In.”

Watch for the next blog entry in about two weeks time! Thanks for reading, and your comments are most appreciated!

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Installing FreePBXv3 in Elastix-2.0 RC3

admin — Tue, 06 Jul 2010 09:53:59 +0000

Yes it’s surprisingly easy, and yes you too can have it today!

This is still very much a work-in-progress, as we’re merging two unfinished products and I’ve not had enough of a chance to test it thoroughly either so I’m hoping for some response from the community on this one. I think this is quite possibly the first time that FreePBXv3 has been tied into a major distro, and I reckon it’d be cool to see Elastix-2.0 support FreePBXv3 officially.

I have however tested Ext’s registering happily, calling each other, so I know that FreePBXv3 is tied in correctly.

Anyway, I’ve tar’d up a slightly older version of FreePBXv3, as they’re undergoing some design changes right at the moment which mean that the current version doesn’t install. So, we’ll be using this.

Total installation should only take about 5 minutes, so this How-To will be nice and short!

So what you’re going to do is start by installing the latest Elastix-2.0 RC3. In my case I used i386.

Once it’s installed, ssh into the server and run:

cd /var/www/html
wget http://c2s.co.nz/dumpfiles/freepbxv3.tar.bz2
tar xvjf freepbxv3.tar.bz2
cd freepbxv3
sh preinstall.sh

It’s 8.5MB, and this is the snapshot of FreePBXv3 that we’re going to be using.

I’ve already modified the preinstall.sh so it uses the user “asterisk” instead of “nobody”, keeping in line with how Elastix does things.

You’re going to then run the following commands to upgrade PHP:

yum install yum-priorities
cd /etc/yum.repos.d/
wget http://dev.centos.org/centos/5/CentOS-Testing.repo
yum --enablerepo=c5-testing install php
/etc/init.d/httpd restart

We need PHP-5.2 because of one of the FreePBX dependencies (Kohana) and Elastix only comes with PHP-5.1

Now you need to fire up the Elastix WebUI, but instead of just going to the regular URL, add in /freepbxv3/ to the end and you should be greeted with the install window:

Agree, hit Next.

You will be prompted with a pre-flight checklist. It should pass the lot. Again, hit Next.

Now, on the next screen, you’re going to see the database details.

Set the database to freepbxv3, the username to “root” and put in the default Elastix root MySQL password: eLaStIx.2oo7

You might also want to adjust your timezone.

Now, create a Username and a Password that you will use to access the system. Whatever you enter in here will be the email address used to log you in, it’ll create the account based off this, so remember the address / password you use. Don’t worry, this account is unique to your Elastix / FreePBXv3 installation.

On the next screen you need to change it from the FreeSWITCH driver to the Asterisk driver. All the other defaults are fine except the AMI admin password which needs to be changed to “elastix456″

The next screen will complain a little about a few bits n pieces relating to Media Management. We’re going to ignore this for now simply because I’m lazy. Scroll right the way down and hit “Next”, as the default selections suit us fine.

It will then go through and do the installation of everything:

When that’s done you’ll be taken to your newly installed FreePBXv3 system!

One thing I’ve found is that the default passwords didn’t seem to work until I modified the password on the accounts. You can do this under Destinations –> Devices

Select Device 1 and choose Edit. Adjust the password and then hit Save.

You should now be able to take a phone and register it using the Username “sip1″.

I then did the same for Device 2, registered with “sip2″ after changing the password, and was able to make calls freely between them.

Granted there’s a lot more to be tested, but that should be enough to get you pointed in the right direction. I’d love to hear from some people how they get on with this, and the blog will be updated in-future.

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New to Asterisk? Learn how to get started today!

admin — Thu, 01 Jul 2010 20:23:04 +0000

Watch this new video from Digium to learn how you can get started with Asterisk! During the presentation you will learn:
What is Asterisk?
* What can I do with Asterisk?
* Which version should I use?
* How do I get started?

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Asterisk 1.4.34-Rc1 And Asterisk 1.6.2.10-Rc1 Now Available

admin — Tue, 29 Jun 2010 10:03:46 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced release candidates of Asterisk for versions 1.4.34 and 1.6.2.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

These release candidates address issues that were reported by the community and resolved since the last round of bug fix releases.

The following is a sampling of issues resolved in these release candidates:

If MeetMe is configured to use dynamic conference numbers, then the first caller (which creates the conference) had to enter the PIN number twice.
(Closes issue #15878. Reported by shawkris. Patched by pabelanger)
Send AgentComplete manager events in the event of blind and attended transfers.
(Closes issue #16819. Reported and patched by elbriga)
Correct manager variable ‘EventList’ case.
(Closes issue #17520. Reported and patched by kobaz)
If there is realtime configuration, it does not get re-read on reload unless the config file also changes.
(Closes issue #16982. Reported and patched by dmitri. Tested by atis)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.34-rc1
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10-rc1

Issues found in any of these release candidates should be reported to the Asterisk issue tracker at https://issues.asterisk.org

Thank you for your continued support of Asterisk!

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Asterisk 1.4.33.1 Released

admin — Tue, 22 Jun 2010 13:04:18 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the release of Asterisk 1.4.33.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.4.33.1 resolves a regression involving the use of FXO signaling in chan_dahdi where a channel could continue ringing after it has been answered.

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33.1

Thank you for your continued support of Asterisk!

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Asterisk 1.4.33 Now Available

admin — Fri, 18 Jun 2010 21:03:28 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the release of Asterisk 1.4.33.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.4.33 resolves several issues reported by the community, and would have not been possible without your participation. Thank you!

The following are a few of the issues resolved by community developers:

Remove arbitrary size limitation for hints
(Closes issue #17257. Reported, patched by tim_ringenbach)
Fix incorrectly typed indications for [nz] stutter and dialrecall
(Closes issue #17359. Reported, patched by alecdavis)
Make AgentComplete message more consistent
(Closes issue #15638. Reported, patched by elbriga)
Missing fallback to audio fax feature when T.38 re-INVITE failed
(Closes issue #16692. Reported, patched by vrban)
Don’t hang up on a queue caller if the file we attempt to play does not exist
(Closes issue #17061. Reported by RoadKill)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33

Thank you for your continued support of Asterisk!

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Asterisk 1.6.2.9 Now Available

admin — Fri, 18 Jun 2010 21:03:25 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the release of Asterisk 1.6.2.9.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.9 resolves several issues reported by the community, and would have not been possible without your participation. Thank you!

The following are a few of the issues resolved by community developers:

Fix the PickupChan() application
(Closes issue #16863. Reported, patched by schern. Patched by cjacobsen.
Tested by Graber, cjacobsen, lathama, rickead2000, dvossel)
Improve logging by displaying line number
(Closes issue #16303. Reported by dant. Patched by pabelanger. Tested by
dant, pabelanger, lmadsen)
Notify CLI when modules are loaded/unloaded
(Closes issue #17308. Reported, patched by pabelanger. Tested by russell)
Make the Makefile logic more explicit and move the Snow Leopard logic down to
where it’s not executed on non-Darwin systems
(Closes issue #17028. Reported by pabelanger. Patched by seanbright,
tilghman. Tested by pabelanger)
Manager cookies are not compatible with RFC2109. Make that no longer true.
(Closes issue #17231. Reported, patched by ecarruda)
With IMAP backend, messages in INBOX were counted twice for MWI
(Closes issue #17135. Reported by edhorton. Patched by ebroad, tilghman)
Fix possible segfault when logging
(Closes issue #17331. Reported, patched by under. Patched by dvossel)
Fix memory hogging behavior of app_queue
(Closes issue #17081. Reported by wliegel. Patched by mmichelson)
Allow type=user SIP endpoints to be loaded properly from realtime
(Closes issue #16021. Reported, patched by Guggemand)

Additionally, the following issue may be of interest:

Fix transcode_via_sln option with SIP calls and improve PLC usage
(Review: https://reviewboard.asterisk.org/r/622/)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.9

Thank you for your continued support of Asterisk!

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The IVR Clinic — with Allison Smith

admin — Mon, 14 Jun 2010 14:44:09 +0000

Allison Smith, The Asterisk Voice

The 15 Commandments of IVR

Commandment #1: Don’t Overestimate Your Listener’s Attention Span

We’re embarking on a fifteen-part-series of blogs designed to assist in the writing of smooth-flowing, efficiently-working IVR systems — as the voice of Asterisk, I’m hoping that my experiences with recording IVR prompts daily will provide some insight into what I consider to be the pitfalls which leads to awkwardly-running and ineffective IVR systems, and ways to avoid them. Many purchases and re-sellers of Asterisk are frequently thrown into the position of having to draft IVR messages for themselves or their clients — hopefully this blog will facilitate the writing of clean, easy-to-navigate prompts.

The very first commandment we’ll delve into: Don’t Overestimate Your Listener’s Attention Span is perhaps the pivotal and most important one of all — a good place to start. If you accomplish this one — it will be easy sailing going forward.

The number one mistake that most people make — when sitting down and coming up with a script that will greet their company’s callers — is over-informing in the opening “main” prompt. Packing way too much information and detail into what should be a concise, helpful directory which sole function is to welcome their callers and sort their request into the appropriate department. I encounter opening greetings almost every day which sound a little like this: (copied verbatim from an actual script. The names have been changed):

“Thank you for calling ABC Grommets — the award-winning grommets you’ve read about in Time and the Economist! Are you constantly disappointed in your grommet choice? Are you continually replacing grommets purchased from that *other* company? You’ve called the right place. Our grommets are 100% nickel, American made, and come in the widest spectrum of sizes and weights in the business. Your satisfaction is our complete focus. We pride ourselves on not only a superior product, but also the best service possible. We look forward to giving you the same care and attention that we’re famous for. If you know your party’s extension, please enter it now. Otherwise…”

They’re kidding, right? They’re actually wanting to make customers listen to that gigantic welcoming speech — and this is *before* they’ve even been offered a list of extensions to choose from? Save the informational, “sales-y” content for your on-hold program. Don’t forget — it’s likely that customers who are cold-calling already know about your company via your website, and are quite possibly calling as a second-string of contact. By all means, confirm that they have the right ABC Grommets on the line (“Thank you for calling ABC Grommets — the #1 supplier of grommets to the Eastern Seaboard…”) but then, immediately start the process of dividing callers into extensions. Think about someone who calls your company multiple times, and how irritating it would be to have to listen to that entire, lengthy commercial more than once. Defer to people’s time constraints (and patience levels) and keep your opening greeting as concise, grabby, interesting-without-being-wordy, and as humanely short as possible.

Keep it simple. Keep it short. Resist the temptation to use your main greeting as a way of dazzling customers or overloading them with information now that you have them “cornered”. Impart only the basic amount of information to set the tone and to best shuffle your customers to the appropriate department — and never forget that the purpose of a good opening greeting is to organize your callers to the right department, so that they may be best served, and your staff’s time is spent most effectively. Plain and simple. Keep this maxim about brevity — and attention spans — in your mind as you draft *all* the options in your phone tree, and you’ll have a smooth, succinct system that’s a joy to navigate around. OK, maybe not a “joy”. But not something people dread.

It might help to remember *your own* last frustrating time spent on the phone when you were trying to work your way painfully slowly through an IVR. I can almost promise you that a lot of that powerless feeling you encountered came from having to listen to too much, and having to part with even more precious time.

Stay tuned for the next installment of our “IVR Commandments” series, where I expose another time-waster: “#2: Thou Shalt Not Create Fake Mailboxes”.

Thanks for reading! Feel free to comment. And watch for the next installment in about two weeks time!

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Asterisk 1.4.33-rc2 and Asterisk 1.6.2.9-rc3 Now Available

admin — Thu, 10 Jun 2010 09:43:56 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced new release candidates of Asterisk for versions 1.4.33 and 1.6.2.9. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

These releases resolve two issues reported by the community:

Fix Debian init script to not use -c.When using the init script as-is currently, it could cause issues on Debian such as high CPU usage. This fix has worked for several people so I’m implementing the change. We now handle color displays properly.
(Closes issue #16784. Reported, tested by pabelanger. Patched by tilghman)
Ensure signals are not blocked inside other signal handlers.This eliminates the annoying on the console.
(Closes issue #17477. Reported by jvandal. Patched by tilghman)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33-rc2
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.9-rc3

Issues found in these release candidates should be reported to the Asterisk issue tracker at https://issues.asterisk.org

Thank you for your continued support of Asterisk!

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Asterisk 1.6.2.9-Rc2 Now Available

admin — Mon, 07 Jun 2010 08:46:07 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced a new release candidate of Asterisk for version 1.6.2.9. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

This release candidate resolves a crash in DTMF detection. See issues #17371 and #17474 for more information.

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.9-rc2

Issues found in this release candidate should be reported to the Asterisk issue tracker at https://issues.asterisk.org

Thank you for your continued support of Asterisk!

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Asterisk 1.4.33-Rc1 And Asterisk 1.6.2.9-Rc1 Now Available

admin — Wed, 02 Jun 2010 10:30:30 +0000

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced release candidates of Asterisk for versions 1.4.33 and 1.6.2.9. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

These release candidates address issues that were reported by the community and resolved since the last round of bug fix releases.

The following is a sampling of issues resolved in these release candidates:

Allow compilation on Mac OS X 10.4 (Tiger)
(Closes issue #17297. Reported by jcovert. Patched by tilghman)
Fix issue where IMAP backend would count messages twice in INBOX for MWI.
(Closes issue #17135. Reported, patched by edhorton. Patched by tilghman)
Fix segfault on logging.
(Closes issue #17331. Reported, tested by under. Patched by dvossel)
Fix crash when processing Cisco DTMF samples.
(Closes issue #17248. Reported by falves11. Patched by dvossel)
Fix crash in check_rtp_timeout.
(Closes issue #17271. Reported, patched by under. Tested by dvossel)
Fix transcode_via_sln option with SIP calls and improve PLC usage.
(Patched by mmichelson. Reviewboard https://reviewboard.asterisk.org/r/622/ )

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33-rc1
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.9-rc1

Issues found in any of these release candidates should be reported to the Asterisk issue tracker at https://issues.asterisk.org

Thank you for your continued support of Asterisk!

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