Comments on: Newbie’s Guide to TrixBox 1.1 and freePBX

By: Luis Gonzalez

Luis Gonzalez — Sun, 10 Oct 2010 13:40:42 +0000

This is one of the most complete guide for trixbox I have seen so far. Great work.

By: Ken

Ken — Thu, 19 Oct 2006 16:45:11 +0000

Hello.
I have installed 3 trunks, all with the same company, Unlimitel. I would like specific extensions to use specific trunks and routes so I can keep track of calling logs. How do I make sure that if one of my customers makes a call, it gets routed to their specific trunk, and proper inbound and outbound routes?
Thanks.
Ken.

By: Tony Davis

Tony Davis — Mon, 02 Oct 2006 19:40:23 +0000

Ward, I have been running a@h for the last 6 months and I am now upgrading to trixbox 1.1. However, the best part of this article is your comment to the type A males. I can attest to the fact that my wife uttered those exact words only days after my first install. They eventually get used to it!!

By: Jim

Jim — Mon, 18 Sep 2006 11:28:45 +0000

A network Sniff revealed the server is just not responding to the sip registration requests. Additionally, after performing an Amportal stop/start, all sip registrations came online and the zap channels went offline. If I reboot,the issue flipflops back to the original. There seems to be a process conflict. What recommendation would you make, use TB 1.1 or return to A@H 2.8 (my current starting point)?

By: Jim

Jim — Sun, 17 Sep 2006 20:53:46 +0000

Ward how are you making out with Trixbox 1.2?? This is the fourth version of a@h/trix… I have installed and it is the first time I’m fairly stumped.. The server does not seem to respond to SIP registrations. IAX and ZAP came right up. So far I have not seen any information regarding this version. I now needto sniff the line to identify who the culprit is. any idea?

[WM: Bad news here, too. 1.2 completely breaks VMware on most boxes, and it’s hit-and-miss on stand-alone machines. We’re gonna pass on this one.]

By: victor

victor — Tue, 12 Sep 2006 18:19:46 +0000

Ward,
I need to add a pause into dial pattern of outbound route – asterisk@home allowed me to have 9|wwXXXXXXXXXX ‘w" for half a second pause. I need this because my trixbox connects to Nortel PBX first before going to pstn. Trixbox dial pattern config doesn’t allow me to add "w", I tried adding it to extensions_additional.conf but it seems that it doesn’t do anything, any idieas?
thanks for all your help

By: Frank

Frank — Thu, 07 Sep 2006 03:26:54 +0000

Ward,
I did as you instructed us to do. I clicked to goto the Online Module Respository and this is what I got?

Terminate Connection to Online Module Repository
Warning: file_get_contents(http://amportal.sourceforge.net/modules-2.1.xml): failed to open stream: No route to host in /var/www/html/admin/page.modules.php on line 445

What do I need to do?
Ward, Thank You! for all you do for us.

[WM: You don’t have Internet connectivity on this machine. Start over.]

By: Tricky

Tricky — Tue, 29 Aug 2006 02:10:52 +0000

Is there anyway to tie it in with vonage? Or do i need to put a digium card in the pc?

Thanks in advance,

Tricky

[WM: Just get an SPA-3000 and search this site for the tutorial.]

By: Andrew Mc

Andrew Mc — Wed, 16 Aug 2006 04:48:24 +0000

I have Sirrix ISDN cards which require the /usr/include/asterisk folder. Trixbox uses CVS code which doesn’t have these folders and the driver install fails. AAH 2.8 looks to use regular release code which does. I don’t know who to tell to look at having these folders put back in the build, but you seem to know so perhaps you could tell them for me.
I have tried puting them in and making some links (trixbox forum) but the update script breaks it again.

By: lockdog

lockdog — Sun, 13 Aug 2006 23:54:12 +0000

great intro.. trying to connect a voiceblue cellular/VoIP gateway as a trunk.. having registration problems.. getting SIP/404 errors.. can sometimes get in as an unknown peer.. any ideas

By: tphank

tphank — Thu, 10 Aug 2006 18:12:30 +0000

Ward: In the past you instructed to do the "yum -y update" after the trixbox update. Should this still be the case? [WM: The TrixBox update script now takes care of this.]

By: Frank Daley

Frank Daley — Wed, 09 Aug 2006 02:27:00 +0000

Awesome intro to Trixbox.

Now trying to get fax-to-email working. Anyone know of a guide for Trixbox 1.1?

ta
frank

By: summers311

summers311 — Fri, 04 Aug 2006 07:27:27 +0000

Hey Ward! Things are a bit weird here. I guess there was another kernal update and it makes things a little screwy. Its 3:00 in the morning on 8/4 here and I am setting up trixbox at my company and am faithfully following your directions as usual, what happens is when you log in the first time it gives you the option of 2 kernals I think one that ends EL and one ELsmp, it defaults to ELsmp and if you follow your instructions in ELsmp it will not load zaptel no matter what you do, but if you make sure to stay in EL and do not do yum updates just trixbox-update.sh I was fine, just thought I would give people a heads up on this and would love to know what the difference is between EL and ELsmp if you know. Can?��Ǩ�Ѣt wait until your next post hope you update TeleYapper soon im a big fan of that one?��Ǩ��.

[WM: Multiprocessor machines (SMP) still are a problem. Andrew doesn’t have one to test. Updates for all our apps are coming … starting next week with AsteriDex.]

By: Scott

Scott — Tue, 01 Aug 2006 01:05:17 +0000

Thanks for all the great articles. I have been using Telasip for 8 months now and love them. One quick note to clear any confusion…Telasip uses multiple gateways so the hostname may not be gw3.telasip.com. It might be gw4.telasip.com, gw5.telasip.com, gw6.telasip.com, etc. Regardless of the gateway, their support staff will tell you which gateway your DID is assigned to use.

By: Sean

Sean — Wed, 26 Jul 2006 04:25:51 +0000

I am running trixbox 1.1.1 and followed the setup above. Most everything works except the Linksys PAP2T-NA. I have a uniden phone plugged into it and I can make calls out. When a call come in th euniden does not ring and does not show caller id. I hear another phone ringing so I pick up the uniden and the call is there and I can talk fine. So it looks like the pap2t is not sending anything over to the uniden. The amount of options in that thing is crazy so I really have no idea where to even start. Has anyone seen this before?
Thanks
Sean

By: Daniel

Daniel — Wed, 26 Jul 2006 02:29:14 +0000

You may want to change your command to copy the ssh keys to the following: cat ~/.ssh/id_rsa.pub | ssh xxx.xxx.xxx.xxx "cat >> ~/.ssh/authorized_keys" If you use the code you have listed, it will overwrite the previously authorized keys, causing confussion, heartburn, and possibily headaches.

By: Maceo Woodward

Maceo Woodward — Tue, 25 Jul 2006 20:50:34 +0000

Does anybody know why the text coloring that existed in Asterisk@Home was ditched in Trixbox? Just curious…

Maceo

By: Tom

Tom — Tue, 25 Jul 2006 15:56:29 +0000

Don’t forget that the Sugar CRM has an admin account with the password "password". There are probably several others that we need to find, since you don’t want the bad guys getting any access to exploit your box.

By: John

John — Fri, 21 Jul 2006 07:50:18 +0000

Incredible! So much work for a free product….. Incredible.
Oops.. the "install-pdf" doesn’t work now… I can’t find any info in the forums, but maybe I’m not looking hard enough.

By: Keen

Keen — Fri, 21 Jul 2006 04:57:54 +0000

Hi, great tutorial! I’m still running an old A@H, will be upgrading once I find the time to. However there’s something that hangs at the back of my head. Currently I have 1 X100P clone FXO card, and I’m thinking of getting another FXO, so that I can dial home from my mobile, and initiate the trixbox to callback to either give me a prompt to use the 2nd FXO to call out to any landline/mobile, or some way of telling the box to dial a number and callback to connect both sides. Is this possible? Would you be able to write a guide/tutorial on this?

Many thanks for your response in advance.

Thank you.

[WM: There are a number of articles already. Just search this site for DISA.]

By: Iain

Iain — Thu, 20 Jul 2006 23:26:48 +0000

Wondeful tutorials, I find them very interesting and easy to follow.

A question: what is the most stable version of Asterisk@Home/Trixbox?
I prefer stability to leading edge.

Asterisk 1.5 used to be great
What about the 2.x series of Asterisk 2.7, 2.8
which one has most things that work 🙂

Thank you again Iain

[WM: All of the AAH and TrixBox versions were built on top of each other. At this point, it’s an easy call. TrixBox can be automatically updated. Asterisk@Home can’t. So, for once, the bleeding edge is probably the most stable.]

By: Roberto Ocampo

Roberto Ocampo — Thu, 20 Jul 2006 06:08:28 +0000

I have record in some mp3 files with all of the sounds from a list of text for the IVR for Asterisk. File loaded from Digium web site. But I don’t know what to do next. Can you tell me a way to convert them to a g729. I have read that this is the better codec. Is it posible? The list of texts were in English and i’ve translated them to Spanish-Mexico. I thik i will be a good contribution. Can you tell me a way?

By: Ian

Ian — Wed, 19 Jul 2006 21:29:44 +0000

I am running trixbox 1.1.0 behind a NAT firewall. I signed up for a voipexpress account from a single incoming DID. I followed your example for the setting up the trunk. It only worked after I changed the context from context=from-internal to context=from-trunk. With the context set to from-internal I would just get a not available tone and the asterisk responds SIP/2.0 404 Not Found. Not sure if this is a bug or a feature.

By: Mike

Mike — Tue, 18 Jul 2006 12:24:13 +0000

Ward – where are you?? We need more. Hope you have settled in after the big move!!! Thx for all of the help!!

[WM: All settled. We’re at pawleys.org until tomorrow. MailCall for Asterisk (listen to your email from any phone!) also coming tomorrow with a little luck.]