Comments on: The Next Frontier: Introducing Asterisk@Home 2.8 and freePBX

By: xlnc22

xlnc22 — Tue, 02 Jan 2007 11:29:32 +0000

A@H on VMWare suffers from timing problems because VMware lacks a reliable timing source – results in a ‘sweaky’ IVR Receptionist!
Has anyone solved this problem?

[WM: Read all the comments to our latest TrixBox 1.2.3 VMware article (option at top of the page). All of the solutions are addressed… and they work.

By: Mark Wallis

Mark Wallis — Wed, 05 Jul 2006 20:39:55 +0000

Having installed the 2.8 ISO and then run:
cd /usr/src
rm -rf /usr/src/freepbx
svn co https://svn.sourceforge.net/svnroot/amportal/freepbx/trunk freepbx
cd freepbx
./install_amp –force-version=2.0.1

I get many lines of:

Warning: fread(): supplied argument is not a valid stream resource in /usr/src/freepbx/install_amp on line 336

Warning: feof(): supplied argument is not a valid stream resource in /usr/src/freepbx/install_amp on line 335

This is doubly strange since it has worked previously with exactly the same h/w. Any ideas out here please?

[WM: Install TrixBox 1.1 instead of AAH 2.8.]

By: William King

William King — Thu, 18 May 2006 00:51:54 +0000

My name is William and I work for the company PC Pharmacy located in Seattle. We just pulled out 40 Dell Optiplex’s GX110. They all have P3 667; 256 MB RAM; 20 GB Hard drive; builtin 10/100 ethernet detected by A@H; built in video; CD-Rom and 2.5 floppy; and beige tower case. I will sell them for $50 plus shipping. I am using one of them now for my business A@H server. If you are interested either e-mail me at william at pcpharmacyrx dot com or call 206-973-3122.

By: Randi

Randi — Tue, 16 May 2006 19:25:17 +0000

THis has been a great site, with tons of help getting my AAH up and running. Question what do I need to watch for if I try to upgrade FreePBX after having already got system up and running. Rushed into the update to 2.8 before article came out. Please help.

[WM: Just follow the freepBX upgrade steps in the article. Oughta work just fine.]

By: Bill Smith

Bill Smith — Tue, 16 May 2006 10:20:42 +0000

I followed the 2.8 instructions and inbound Telasip go un-answered, the calls do not even hit my AAH box.. Has there been a Context change (I use from-pstn – per Telasip) or some Trunk Settings??

Thanks!!

By: youngwax

youngwax — Sat, 13 May 2006 01:23:05 +0000

To get imap support in php for the asterisk@home/ centos / sugarcrm package, try (command line)
yum install php-imap
I rebooted after that. I think that’s all you need

By: Bruce

Bruce — Thu, 11 May 2006 08:33:03 +0000

I too am not able to recieve inbound Telasip calls but out bound work fine.. Has anybody solved this issue???

Thanks!

By: AV

AV — Wed, 10 May 2006 05:03:17 +0000

Quote: As tempting as it may be to make changes to the internal code of freePBX, DON’T!
So my question: is this applicable to CONF files, for example sip_nat.conf?

By: Robert Moskowitz

Robert Moskowitz — Tue, 09 May 2006 12:40:09 +0000

Question on zaptel rebuild:

Your 2.7 instructions includes a patch for spinlock. But with 2.8 we are on CentOS 4.3. Is this patch still required? If so is the source still in the same directory?

Thank you for your wonderful set of instructions. I and many others know to look here. Is there at least a pointer to here in the Asterisk WiKi?

[WM: Spinlock patch is not required in 2.8.]

By: Robert Moskowitz

Robert Moskowitz — Mon, 08 May 2006 17:50:05 +0000

You state that Internet connectivity is needed for the install to succeed? Why? Where I am installing the system, I do not have DHCP. I know how to get in on console and mode the necessary files to bring the interfaces up afterwards. But if the install had to go out and fetch something, I missed it.

I am proceeding with the yum update, but if I have to get a DHCP server on that segment and reinstall, please tell me, and update your ‘internet connectivity’ to be a little more helpful (interfaces need manual config or not all software installed).

[WM: Robert, it sounds like you know a lot more about this than we do. Our technical explanation is "ours breaks" without Internet connectivity. There is undoubtedly a way to fix it … as with all things Linux. We just don’t know how. So our advice is to set up Internet connectivity first. Then it just works, and you don’t have to wrestle with it. Good luck though!]

By: Mike Watters

Mike Watters — Sat, 06 May 2006 16:54:46 +0000

Well I upgraded to 2.8 and find it to be a pleasurable experience. However, I now cannot recieve inbound Telasip calls. It seems with original settings copied from 2.7 inbound calls hit my box but are answered by Telasip and default to my email at thier end. Any ideas?

By: Brodie

Brodie — Sat, 06 May 2006 04:37:40 +0000

I’ve been trying to get SugarCRM to get my pop email, but it says I need the IMAP libraries compiled into PHP or something… An article on how to do so would be really cool. I’ve been reading your articles for quite some time now and they’ve been a Godsend.

Thanks for everything guys!

Brodie

By: Mike

Mike — Sat, 06 May 2006 00:22:48 +0000

Ward,

2.8 is great and thanks to you I had it up and running on a new machine within 30 minutes. ZAP auto configured and works without effort. However my Telasip inbound is broke, any ideas?

Mike

By: Antonio Montani

Antonio Montani — Fri, 05 May 2006 23:01:44 +0000

hey! whatever happened to the asterisk info page that gave you status on EVERYTHING! it was so great! cant find it anywhere on freepbx... [WM: Here's the fix for putting the Maintenance Tab back in AAH 2.8 once you upgrade to freePBX 2.1.]

By: Pat West

Pat West — Tue, 02 May 2006 14:35:37 +0000

My solution to Festival is to use flite. Which can be found here: http://cmuflite.org It has been my experience with using the slower machines that Festival needs way too much CPU and that some or most of the delay is the conversion time. The Festival scripts for Asterisk produce a .wav that is played back after conversion. Flite was designed for speed on embeded systems and PDA type devices and uses very little CPU. It's default voice is also setup for 8KHZ which is perfect for Asterisk. It's a very simple tweek (for those who are inclined) to replace text2wav with Flite in the Festival scripts. There is even an app_flight floating around for direct Asterisk interface. See: http://dialogpalette.sourceforge.net/ and check the downloads. [WM: Great tip. In fact, it's so GREAT that we plan to cover it in an upcoming article. Can't get app_flite to speak with AAH 2.8, but Flite works great in an AGI script. If you've had any success with app_flite, drop us a line.]

By: Kaleb Kunzler

Kaleb Kunzler — Tue, 02 May 2006 02:40:44 +0000

Regarding the 7-second delay with festival . . . When I dial *51 to have festival read my weather report, the first time it downloads a new report I get the delay, but if I call a second time within a short period of time (the weather info is already cached) then I don't experience a delay. It seems the "easiest" way to remove the delay would be to have a background task automatically check and cache the weather info regularly, and have the weather feature always use the cached version. [WM: You're correct, but Festival has numerous other uses that are all but worthless with such a lengthy delay. I'd prefer to see the problem fixed so that it works like our AAH 1.5 system still does. Then again maybe we'll just hang on to the 1.5 system for use as a Festival server.]

By: Rob Thomas

Rob Thomas — Tue, 02 May 2006 01:14:50 +0000

They turned to ‘unknown/invalid’ because we changed from a ‘module.ini’ setup to a ‘module.xml’ – We’re Buzzword Compliant! Removing the modules would _probably_ have deleted the contents of the table for each module, and if it did that, it would have stuffed up any previously set up IVRs, time conditions, etc. Correct upgrade procedure is ./install_amp –force-version=2.0.1 (The ‘–force_version’ shouldn’t be required, but it doesn’t hurt to do it), go into modules, download and install all of them in online modules, go into local modules,click install, and upgrade if you’re prompted to do so. Yes, we know it’s a lot of clicking, and 2.1beta2 should fix that up too. (Ooh, I’m letting secrets out now! Ryan told me this morning that he’s done a whole pile of work on the module page, which I’m looking forward to seeing!)

By: Rob Thomas

Rob Thomas — Mon, 01 May 2006 21:43:33 +0000

Couple of minor corrections – you don’t need to uninstall modules before you upgrade them. They all should upgrade themselves seamlessly.
There’s also a known bug in 2.1beta1 that stops the ‘Trunk Dial Paramaters’ from working. That’s fixed in SVN which, if the users follow your instructions, they’ll be using… Umm. I think that’s about it.

[WM: Rob, thanks. After upgrading to 2.1beta1 from the 2.01 version shipped with Asterisk@Home 2.8, all of the installed modules turned to gibberish and appeared to be corrupted. That’s the only reason I suggested removing them first. Once the upgrade to 2.1beta was completed, further upgrades (of which there were about a half dozen this morning) worked as advertised, i.e. no need to uninstall a prior version before upgrading. Since AAH 2.8 users all are building a new system from scratch, it was pretty harmless to delete the problem modules before proceeding just to be on the safe side. But it’s good to know that, in the future, that won’t be necessary. Thanks for reviewing everything. We’ll sleep better now.]

By: Ward

Ward — Mon, 01 May 2006 19:39:03 +0000

There’s still a 7-second delay in reading each line of text with Festival in this new version using our old clunker test machine with 256MB of RAM. This has existed since AAH 2.0. If you get better results, please post them. Otherwise, there’s still a bug somewhere.

By: NerdBob

NerdBob — Mon, 01 May 2006 18:34:11 +0000

Casey, that's the case after you upgrade freePBX to version 2.1beta1, where this is implemented as a feature, rather than missing maintenance tools. Again, this has been discussed a lot, and Rob is working on getting them back in the 2.1 final release, or probably even in the next release that's due toward the end of the week if I am not wrong, having read from the Whirlpool forums. Cheers.

By: Casey

Casey — Mon, 01 May 2006 18:19:55 +0000

Maintenance functions are provided from within the FreePBX structure (at least in the A@H 2.8 ISO). Look in the Module Administration area. You'll find them listed. Once you install and then enable them, you can access them thru the TOOLS menu. No need to fiddle with trying to port them from a previous version. [WM: Once you upgrade to freePBX 2.1, the Maintenance functions disappear. Here's the new fix.]

By: Ben

Ben — Mon, 01 May 2006 18:09:26 +0000

There are few webpages I keep coming back to – yours is one of them. I had a number of issues with Asterisk in the early days, this site helped me solve them.

Good luck with the move, cant wait to read more about what you have found in the system and ways we can make it do more cool things.

By: Dave

Dave — Mon, 01 May 2006 13:03:26 +0000

Yay! No more slash and burn upgrades! (I couldn’t wait, installed 2.8 last week but still love the article)