Comments on: Newbie’s Guide to Asterisk@Home 2.2: Unabridged Soup-to-Nuts Installation Guide

By: 小百合

小百合 — Sat, 02 Jan 2010 02:35:38 +0000

happy new year!

By: charly

charly — Wed, 04 Apr 2007 05:02:58 +0000

HI guys i’m looking a place to find some help or a place to write my problem with asterisk and a SIP/TRUNK. Someone knows where? or who?(i already went to the forums)

By: Ed

Ed — Wed, 06 Dec 2006 04:57:32 +0000

I have installed TrixBox and Asterisk PBX within Windows. My system configuration includes a Celeron 2 GIG processor and 1 Gig of RAM and DSL connection. When testing the voicemail platform I am experiencing the choppy voice quality with internal voice prompts. I have changed various codecs using the soft phone options. I have also tried changing jitter buffer from between 5ms to 100 ms with no improvement. I am not currently with SIP provider, testing internally with IP. Any Thought's into how to minimize choppy voice quality ? [WM: Be sure you're running the latest, stable version of TrixBox (1.2.3) from our build. And, if you still have problems with choppy sound, apply this fix.]

By: Greg

Greg — Sat, 18 Mar 2006 04:38:37 +0000

If you are looking for the UIP-1868 to work with asterisk, take a look at: http://perceiva.com/technologies/docs/uip1868.html

Took a lot of tinkering since it doesn’t come with any directions…but you must get the unlocked version (not the Packet 8 or Vonage version) for it to work with Asterisk

By: Juan Ochoa

Juan Ochoa — Thu, 26 Jan 2006 13:29:24 +0000

Very good tutorials, these are even better than the official How To. I want to know something, you recommend the ip phone Uniden UIP-1868P IP Wireless Phone, I saw in the Buy.com esepecification page, that it only work with Packet8. Can I use it as an extension in my A@H box?

[WM: That particular model is locked to Packet8 so it won’t work with Asterisk. We like Uniden’s wireless phones, however, and the one and two-line 5.8GHz phones work great with an SPA-3000.]

By: Tom Mangiacapre

Tom Mangiacapre — Wed, 11 Jan 2006 14:49:28 +0000

A helpful tip: If you are using TelaSIP and experiencing one way audio, you may want to try a different telasip-gw host entry as the one stated in this (excellent) guide. Instead of the entry in the article, try this in sip_additional.conf:

[telasip-gw]
type=peer
host=gw4.telasip.com
qualify=yes
insecure=very
context=telasip-in
username=yourusername
secret=yourpassword

Doing this solved my one way audio problem. I got this setting from TelaSIP support.

By: Walter

Walter — Thu, 05 Jan 2006 17:12:07 +0000

A couple of things: Problem: Has anyone worked with integrating with Avaya (S8500 w/ Communication Manager in particular, not Definity). I am working on this now, but I cannot get the PBX to open an outbound channel to the Avaya system. Also, is there a debug watch mode that you can enter on Asterisk to watch what happens during a call on the PBX side? Tailing a log file maybe? [WM: Hi Walter, Questions such as the above are better posted on the Voxilla or SourceForge forums where you can get threaded input from numerous individuals. Best of luck.] Additional Components: One thing I found useful (if you are going to add components to this setup) is to add gcc-c++ to this image. The tools to add H323 ability will not compile without this installed. Of course yum -y install gcc-c++ will do the trick for most, I think it should be just included. [WM: Agreed! You might want to pass this along in a new message thread on SourceForge which Andrew reads daily.]

By: Terry Young

Terry Young — Mon, 02 Jan 2006 22:36:41 +0000

Hi Ward

Thanks for all the hard work in producing your Asterisk@Home tutorials – I’d be lost without them.
I’ve been trying to correct the Max Channels bug in my 2.2 asterisk@home installation and in my newbie ignorance I think there could be a } missing from the new s,7 line. Please could you confirm so that I don’t have to jump out of the nearest window. Thanks Terry

[WM: The way it’s shown in the article is what is shown in the bug report.]

By: Paul Martin

Paul Martin — Mon, 02 Jan 2006 02:16:07 +0000

for a good SSH client on WIN32 use PUTTY
http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html

By: Kenneth

Kenneth — Fri, 30 Dec 2005 21:55:30 +0000

Hi and thanks for all the great stuff on AAH.
As a beginner to AAH (just installed V2.2 and have it running nicely) and ip-telephony I am truly grateful for these guides and insights to this GREAT system. I am amazed at what I can do already!!

I am currently rebuilding our phone setup at home and looking forward to getting deeper inside AAH. A thing for the wishlist would be some words on how to strip the aah-setup for things that aren?��?�?Ǭ�?��Ǩ?�?ɂİ?Ǭ�?��Ǩ?�?ɂ��?Ǭ�t ?��?�?Ǭ�?��Ǩ?�?ɂİ?Ǭ�?��?��?ɬ�must haves?��?�?Ǭ�?��Ǩ?�?ɂİ?Ǭ�? for the home user, like A2Billing and maybe the CRM.

Thanks Again.
/Kenneth

[WM: Kenneth, Thanks for the kind words. As for stripping out unnecessary code, do yourself a HUGE favor. Don’t! I don’t know how many people I’ve cautioned over the years about this only to have them do it anyway and then find out they’ve deleted a component that they really needed or the system needed to function reliably. So, one more time, DON’T!]

By: DanITman

DanITman — Thu, 29 Dec 2005 22:48:43 +0000

#18 can you elaborate more. I am having the same problem as #12 but I can’t find a link to read the sourceforge article.

Thanks

[WM: See link in #17 Addendum above. The main article has also been amended.]

By: Astronot

Astronot — Wed, 28 Dec 2005 03:27:27 +0000

I finally figured out #12’s solution. Simpler than it seemed; those two files are missing the zaptel entries after the yum update, but they can be recovered from the rpm saves of the same files. Thanks for your response WM, and happy New Year!

By: Jacob

Jacob — Mon, 26 Dec 2005 19:08:26 +0000

I keep getting the following when using the above NAT address update script. I’m not much of a linux buff, so I don’t know what’s wrong.

no crontab for root – using an empty one
crontab: installing new crontab
"/tmp/crontab.XXXXzGioi0″:2: bad minute
errors in crontab file, can’t install.
Do you want to retry the same edit?

Thanks for your helpful,easy-to-understand blog! Keep up the top-notch work!

[WM: Post the crontab entry so we can have a look.]

By: Astronot

Astronot — Mon, 26 Dec 2005 19:04:10 +0000

I'm having the same problems as #12 up there, not being able to get zaptel working after the yum update. I get an error: "Error: missing /dev/zap!" when it tries to start up zap. I don't understand #12's solution, so if someone wouldn't mind elaborating and/or updating the article, I'd appreciate it. Great work here! [WM: see below. If you're still having problems, take a look at the Digium card discussion in this HOW-TO Guide from Australia.]


yum -y update
reboot
rebuild_zaptel
reboot
genzaptelconf
reboot

Addendum. I tracked down your problem on SourceForge. Here's the discussion thread that will show you how to fix the yum update problem. Sorry, but I don't yet have any Digium cards, or I could have been more helpful.

By: Jules

Jules — Mon, 26 Dec 2005 10:49:24 +0000

A job well done needs to be recognized….I’ll then join the guys here and say "Congrats" for such a great documentation…it really got me going….

A2Billing seems to be interesting…can’t wait for an article on that one!

Again, thanx

Jay

By: Michael Davidson

Michael Davidson — Sat, 24 Dec 2005 08:43:01 +0000

I need to know how to setup H.323 extensions in A@H. I’ve done some reading and it seems like I have to setup some kind of forwarding from the H.323 Gatekeeper to a SIP extension and vice versa to have it work, but this seems a little over complicated for a PBX (*) that supports H.323. Any suggestions would be great!

Thanks,

Michael Davidson

By: Clive Carter

Clive Carter — Sat, 24 Dec 2005 07:40:45 +0000

Thankyou thankyou thanyou !
Thankyou to NerdVittles for 2.2 (as well as Andrew obviously) , and thankyou to Mr Berry. That last problem has been driving me nuts for a couple of days !
Now working ok

By: Joel Berry

Joel Berry — Fri, 23 Dec 2005 22:56:48 +0000

Just starting going through the AAH 2.2 article. I was getting stuck after the yum -y update. I could not get the zaptel devices to show up, even after a kernel rebuild and genzaptelconf. After some digging, I found that one of the rpms that were updated, overwrote the /etc/udev/rules.d/50-udev.rules and /etc/udev/permissions.d/50-udev.permissions files. These files used to contain some zaptel entries that were blown away. Merging the changes back into these files solved the problem. Just a FYI.

By: dstroot

dstroot — Wed, 21 Dec 2005 01:37:45 +0000

One suggestion:

When you edit your vm_include.inc file your can change the link to ARI which I think is a much cleaner more professional interface.

"nano -w /etc/asterisk/vm_email.inc. Press Ctrl-W, type /cgi, and press the enter key. You?��?�?Ǭ�?��Ǩ?�?ɂİ?Ǭ�?��Ǩ?�?ɂ��?Ǭ�re now positioned where you need to type either the fully-qualified domain name for your Asterisk server or the private IP address if you only want to read your emails from behind your firewall."

Instead, use "http://my.fullyqaldomname.com/recordings"

Also – the current version of AAH doesn’t have the latest ARI code. Get it here: http://www.littlejohnconsulting.com/?q=ari

– DJS

By: ip4all

ip4all — Tue, 20 Dec 2005 16:17:25 +0000

Ward, you should publish this great work on A@H as a book. I also have a question to you. I followed the instructions to install a@h 2.2 and now I am not seeing any incoming numbers in the caller ID. I see my own BV number in CID. If I revert back to 1.5 all things work just fine. Any, idea what might be the problem ? Thanks

[WM: Thanks for the book suggestion, but we don’t do books. As for BV, read the BV setup article on this site. BV does things a little differently … to put it nicely.]

By: Brent

Brent — Mon, 19 Dec 2005 15:17:11 +0000

Hey there, great blog. Currently working with @Home 2.1 and have an Analog phone, SIP (X-Lite) phone and Polycom 300 IP Phone associated with my PBX. I can get each phone to call both ways with exception of the Polycom, where I can’t dial out. Wondering if could please point me in the right direction.

[WM: Head over to the Voxilla Asterisk forum or the SourceForge Asterisk@Home forum and someone will get you going in short order.]

By: Richard

Richard — Mon, 19 Dec 2005 13:24:36 +0000

Have you tried Idefisk
http://www.asteriskguru.com/tools/idefisk_beta.php
I?��?�?Ǭ�?��Ǩ?�?ɂİ?Ǭ�?��Ǩ?�?ɂ��?Ǭ�ve found it to be excellent and wonder whether it might be worth introducing it to your following?

I cant wait until your presence article – is it going to be Jive/Wildfire based?

Great blog!!!

Rich

By: jochen

jochen — Mon, 19 Dec 2005 11:33:22 +0000

I am facing a lot of problems after moving from 1.3 to 2.2, but I have to thanks too for this well written guidance. Maybe you can help?

I am from Germany, which means e.g. I can?��?�?Ǭ�?��Ǩ?�?ɂİ?Ǭ�?��Ǩ?�?ɂ��?Ǭ�t see [de] in the indication.conf, etc.
But some more important question:

1. recording:
How I have to configure the recording feature for each chat (record in/out in extension and General settings/Dial command options). I can?��?�?Ǭ�?��Ǩ?�?ɂİ?Ǭ�?��Ǩ?�?ɂ��?Ǭ�t find the voice files in the defined directory (agents.conf: savecallsin=/var/calls) or typical directories like var/lib/ or var/spool/ (the mentioned monitor directory does not exist).
How about urlprefix= and createlink=, nothings works
2. big echo problems (in contradiction to 1.3):
How I can improve/fix this problems? I have no idea what to do first.
3. Queue problems: HoldMusicCategory: the mentioned directory is filled with mp3 standard files, but no music is played during the waiting period
Also there is no Caller Announcements for the first caller in the queue, only the second caller gets an announcement.
Also the ring on one of the phones (ringall) stops after 20s, why ?��?�?Ǭ�?��Ǩ?�?ɂİ?Ǭ�?��Ǩ?�?ɂ��?ɬ? this is maybe easy if I can change the ?��?�?Ǭ�?��Ǩ?�?ɂİ?Ǭ�?��?��?ɬ�retry?��?�?Ǭ�?��Ǩ?�?ɂİ?Ǭ�?-Select-Button in queue definition? Where I can find this page?
I am also not sure if the wrap-up-time is working correct, because (I defined ?��?�?Ǭ�?��Ǩ?�?ɂİ?Ǭ�?��?��?ɬ�0?��?�?Ǭ�?��Ǩ?�?ɂİ?Ǭ�?) the next 20s should ring in this case immediately after the first try.
4. voice-message e-mail-delivering:
to send voce messages isn?��?�?Ǭ�?��Ǩ?�?ɂİ?Ǭ�?��Ǩ?�?ɂ��?Ǭ�t possible (but this is not easy to explain)

I am using only external IAX-trunks and internal SIP phones (hard + soft)
Thanks in advance for any advise.

[WM: Sorry but we don’t do tech support. I’m retired, remember! You might want to visit the Voxilla forums and post some of your questions there. I’d do them one at a time if you expect a response. Good luck!]

By: Amado

Amado — Fri, 16 Dec 2005 22:09:45 +0000

I recently followed your guide for Asterisk@Home 2.1 and everything went great, I have a working PBX. Now I see version 2.2 has come out. Is there a way for a Linux and Asterisk newbie to upgrade version 2.1 to 2.2 without starting over? [WM: There's really not an upgrade path that works. Lucky for you it's only been a week so write down what you've got, make copies of all the files in /etc/asterisk (but use these for reference ONLY; don't just copy them into 2.2 or you may get unpredictable results!), and print screen shots of all the Asterisk@Home AMP screens to give you an idea how you''ve set things up. Trust me. It's a lot easier the second, third, fourth, fifth, ... well, you get the idea.]