Comments on: ISP-In-A-Box: Installing a Free Asterisk PBX Phone System (Part II)

By: Octothorpe

Octothorpe — Mon, 18 Sep 2006 00:05:14 +0000

In reference to #20 above, I always have just downloaded the ISO and burned it with NERO with absolutely no problems, other than the ocassional bad burn. You do need to make sure that your PC bios is set to boot off of the CD or no CD will appear to be bootable.

[WM: Ditto.]

By: Charles R Downs

Charles R Downs — Sun, 17 Sep 2006 16:00:50 +0000

On the question #18. Just burning the ISO on the CD is not enough. You have to make the CD a bootable CD. Nobody explains this. I worked very hard using Nero to get the CD to be bootable. It took several tries.

By: Ruben

Ruben — Sat, 16 Sep 2006 01:29:50 +0000

Has anyone setup their asterisk system for TelaSip. I need to set up the inbound, and outbound trunks and dial plans. I have inbound working from the free sip provider that I read about in NerdVittles. Now I 'paid' for TelaSip and can't configure it correctly. I've read everything I can get my hands on online. Can an expert help? Thanks Ruben. [WM: Our site is full of configuration information on most providers including TelaSIP. The full list is here and TelaSIP is here.]

By: Dave

Dave — Mon, 07 Aug 2006 19:14:00 +0000

Your tutorial was great, confirmed what I had figured from 2 other tutorials, tinkering, and docs from broadvoice tech support. Have had outbound working fine for days but COULD NOT get inbound working. Finally solved it:
1) log into broadvoice and turn off voicemail (their own people tell
you this.
2) when receiving an incoming call I would watch the SIP DEBUG output
and would always get "407 Proxy Not Authorized". FINALLY
figured out that in the registration line the second part SHOULD
NOT BE "sip.broadvoice.com" but the broadvoice proxy server, here’s
my register line (passwords removed):

9125551212@sip.broadvoice.com::9125551212@proxy.nyc.broadvoice.com

Hope this helps. Also, th elast part of your article was really helpful in that ASTERISK IS CONFIGURED BY DEFAULT to route inbound SIP calls to "congested" and then hangup. This needs changed as you describe and then it works great!

By: ali

ali — Mon, 26 Jun 2006 22:03:27 +0000

i have downloaded and burn cd for Trixbox. When i reboot my PC with this cd in it, instalation does not start. Tried couple times no luck!!!!
any word of advice??

[WM: Probably a bad ISO image. Try another download from a different site.]

By: Mike

Mike — Thu, 15 Jun 2006 05:29:20 +0000

Great work. Two Questions, do the tutorials for asterisk@home port over to trixbox? I am having trouble making outbound 800 calls with free world dialup. I am registered with FWD in the asterisk_info page. I can connect for about 9 seconds when I hangs up. Any help wold be greatly appreciated. [WM: Read the current articles on Nerd Vittles for TrixBox tutorials. As for FWD and 800 calls, we've always had mixed luck with FWD. Sometimes it's great. Often it's not.]

By: Ian Worthington

Ian Worthington — Mon, 20 Mar 2006 14:29:38 +0000

"let me admit that we use numbers in the range of 200 to 399″.

Numbers in the range 3xx are speed dial numbers — not good for extensions unless you want to get very confused!

By: Bobby Feather

Bobby Feather — Sat, 11 Mar 2006 05:03:13 +0000

I might be able to help with the t1 card problem. I have never used one of the cards before but i work for a Telcom copany as a tech and work with t1’s ever day. Setting up the card does not sound to hard to work out. Send me the type of card and the model # and i will see if I can find something and point out the main options to set.
braintumor2000@gmail.com

By: Paulo Durano

Paulo Durano — Fri, 03 Feb 2006 16:08:21 +0000

Outgoing call is ok. Incoming says "busy". Pls. help me solve the problem. Thanks
Incoming Settings

context=from-pstn
dtmf=rfc2833
dtmfmode=rfc2833
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
insecure=very
nat=yes
secret=********
type=user
user=7328101969
username=7328101969
Thank you.

[WM: To fix it, choose AMP->Maintenance->Config Edit->extensions.conf->from-sip-external. Comment out all the lines in the existing file by adding a semicolon at the beginning of each line. Then add the following line, save your changes, and reload Asterisk]
exten => _X.,1,Goto(from-pstn-timecheck,s,1)

By: Paul Durano

Paul Durano — Thu, 02 Feb 2006 17:06:03 +0000

I admire people who share knowledge to anybody. This information is highly appreciated. God bless to Free information.

By: Lee

Lee — Fri, 27 Jan 2006 13:42:36 +0000

I’ve been really impressed by the amount of useful information on your site. I have always thought that Asterisk is fantastic but with this level of information it because even better. I have been trying to work out how to implement a follow-me facility but unless I am missing something it doesn’t seem completely straight forward. I will crack on… It would be really great if you could add an article on follow-me using asterisk to allow for hotdesking.

By: Darren

Darren — Thu, 05 Jan 2006 20:43:33 +0000

I am trying to integrate my CCM with Asterisk using a sip trunk. I am able to place calls from asterisk to my CCM servers but I have been able to place calls via CCM to asterisk. Can someone please point me in the right direction, I am new to Asterisk but I have been working with CCM for a long time now. Here is a copy of the error I am getting …

[WM: Hi Darren, Questions such as this are better posted on the Voxilla or SourceForge forums where you can get threaded input from numerous individuals. Best of luck.]

By: Nat

Nat — Fri, 16 Dec 2005 22:03:44 +0000

Your articles have helped me a lot, thanks. However, I've not been able to make or receive calls on our regular phone lines. We have a T1, and I purchased a Digium TDM2412B card for this purpose. There seems to be very little information about this type of setup. I can make outbound calls using voipjet without problem, but when I try to make or receive calls through our T1 phone lines, I just can't. I have a Zap trunk pointing to g0 and as far as I know it should work, but it doesn't. Do you have any idea of what am I missing or where to get this info? Thanks. [WM: I don't have a T1 line or card so I'm not going to be much help. There is lots of information on T1 cards on the Digium site. I'd recommend you post your questions in either the Asterisk-Users listserv or on the Asterisk Users Forum. Search around a bit in both places first. Good luck!]

By: Austin

Austin — Thu, 01 Dec 2005 23:10:50 +0000

Thanks for the article. read it all, it all work except that in sames my incoming call is not getting answer at all or routed to the extension.

here is a debug

— Executing AbsoluteTimeout("SIP/2403990305-9283″, "15″) in new stack
— Set Absolute Timeout to 15
— Executing Congestion("SIP/2403990305-9283″, "") in new stack
== Spawn extension (from-sip-external, 2403990305, 2) exited non-zero on ‘SIP/2403990305-9283’
— Executing AbsoluteTimeout("SIP/2403990305-9283″, "15″) in new stack
— Set Absolute Timeout to 15
— Executing Congestion("SIP/2403990305-9283″, "") in new stack
== Spawn extension (from-sip-external, h, 2) exited non-zero on ‘SIP/240399030

help!

WM: Congratulations. You’ve found the #1 problem with the default Asterisk Management Portal (AMP) configuration. Just edit the from-sip-external context in extensions.conf and comment out all the existing lines. Then add the following line and reload Asterisk:

exten => _.,1,Goto(from-pstn-timecheck,s,1)

By: Ron

Ron — Tue, 22 Nov 2005 06:08:20 +0000

Thanks you saved me on the SPA-3000 install. I would have been still trying to configure it a month from now.
How about universal plug and play everyone??!!

By: Justin

Justin — Fri, 18 Nov 2005 03:03:31 +0000

I followed your instructions to the T. Beleive me. I even completely reformatted and readded all the info and only your info back into the sytem but still I cannot get incoming calls to complete. They come in to the system but the system drops them. See the log file. Nov 17 22:00:07 VERBOSE[1328]: -- Executing AbsoluteTimeout("SIP/5182121068-2d9e", "15") in new stack Nov 17 22:00:07 VERBOSE[1328]: -- Set Absolute Timeout to 15 Nov 17 22:00:07 VERBOSE[1328]: -- Executing Congestion("SIP/5182121068-2d9e", "") in new stack Nov 17 22:00:07 VERBOSE[1328]: == Spawn extension (from-sip-external, 5182121068, 2) exited non-zero on 'SIP/5182121068-2d9e' Nov 17 22:00:07 VERBOSE[1328]: -- Executing AbsoluteTimeout("SIP/5182121068-2d9e", "15") in new stack Nov 17 22:00:07 VERBOSE[1328]: -- Set Absolute Timeout to 15 Nov 17 22:00:07 VERBOSE[1328]: -- Executing Congestion("SIP/5182121068-2d9e", "") in new stack Nov 17 22:00:07 VERBOSE[1328]: == Spawn extension (from-sip-external, h, 2) exited non-zero on 'SIP/5182121068-2d9e' Nov 17 22:00:07 DEBUG[1328]: cdr_mysql: inserting a CDR record. Nov 17 22:00:07 DEBUG[1328]: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2005-11-17 22:00:07','"Unavailable" <147.135.20.128>','14713520128','5182121068','from-sip-external', 'SIP/5182121068-2d9e','','Congestion','',0,0,'NO ANSWER',3,'') Nov 17 22:00:07 DEBUG[1328]: update_user_counter(5182121068) - decrement inUse counter Nov 17 22:00:07 DEBUG[1328]: 5182121068 is not a local user Nov 17 22:00:07 DEBUG[1328]: Stopping retransmission on '1ff008d-a6@147.135.20.128' of Response 1: Not Found Nov 17 22:00:08 DEBUG[1328]: Stopping retransmission on '6c9629b067a8d64073aa98ba4dbecd18@127.0.0.1' of Request 103: Found Nov 17 22:00:08 DEBUG[1328]: Registration successful I tried adding context=from-pstn to both broadvoice entrys in sip_additional.conf and all that does is keep anything from hitting the log file. I get either a busy signal or voicemail if it is enabled. [WM: Read this column (particularly the discussion of from-sip-external), and I think you'll see what's missing.]

By: Roosevelt

Roosevelt — Tue, 25 Oct 2005 04:54:03 +0000

I just wanted to know who is my isp provider and my ip address.

By: marc

marc — Fri, 09 Sep 2005 23:18:27 +0000

thanks for the feedback. i isolated the problem and now outgoing calls
work. unfortunately, there were so many variables, i can’t remember
exactly what i did to fix the problem.

i am now on to the next problem! incoming calls..they go straight to BV
voicemail. i had a number of ideas, but none of them seem to fix the
problem. i turned on debugging and placed a call to my BV number and
below is the results.

AsteriskInfo shows that i am being registered properly with BV.

Here is the debugging info. any thoughts appreciated…thanks.

Sending to 147.135.0.128 : 5060 (non-NAT)
Looking for 2104640055 in from-broadvoice
SIP/2.0 404 Not Found

hi. me again. i was able to figure out my problem. you can follow my struggle here: http://tinyurl.com/8yjxa

but your blog was a great resource. i’ll be back!

[WM: Here’s a good example of how NOT to implement Asterisk. From your post on Voxilla, it’s apparent that you were using three different tutorials to configure Asterisk. Nothing in mine ever mentioned a from-broadvoice context so … The bottom line is try baking the cake according to the directions in one cookbook before you start adding other ingredients. Otherwise, you end up with a debugging mess. mberlant is good at untangling things; however, I’ve worked hard to avoid spending my life debugging other folk’s code, and I’m not inclined to change directions now. But, thanks for sharing your quest.]

By: marc

marc — Fri, 09 Sep 2005 08:24:10 +0000

I too have the same problem Mike did. When I place an outgoing call to my cell phone from my Broadvoice phone I hear "all circuits busy". However the call actually goes through as my cell phone starts to ring! I have rechecked the trunk and outbound routing...but can't figure it out. [WM: Something's wrong ... but you knew that. My guess would be that you're not getting properly registered with BroadVoice or your SPA-3000 is misconfigured. But there's an easy way to find out. Go to the Asterisk console, log in as root, and then start up the Asterisk console: asterisk -r. If you have trouble reading the messages quick enough, you may need to use SSH from a Mac (ssh root@ipaddr of your Asterisk box then asterisk -r) or Putty from a PC. Report back what you find when Asterisk tries to place the call. Another place to look is AMP->Maintenance->AsteriskInfo which will tell you whether Asterisk is getting properly registered with BroadVoice. Finally, use a web browser to check the Info tab of your Sipura SPA-3000 using your Admin login and check the Registration State under Line1 and PSTN. Both should say Registered.]

By: Ward Mundy

Ward Mundy — Sun, 04 Sep 2005 21:18:11 +0000

Password update process was added for Sugar CRM today. Start up AMP. Click CRM. Log in with username admin and password of password. Click on My Account and then update password to change password.

By: Mike

Mike — Fri, 26 Aug 2005 01:27:33 +0000

The night before I googled out your instructions – I had spent several hours, learning a lot, but nonetheless failing at completing an inbound & outbound call 🙁 I was aware of the need for a route – but I had missed something in setting up the trunk… although from the Maint screen – I had registered with the SIP Peer I would always get a trunk busy …. Anyway… I plan to look more into your configs vs mine and LEARN where I went wrong…. THANKS A BUNCH!!!

By: Dave

Dave — Fri, 19 Aug 2005 15:04:03 +0000

This tutorial didn't work for me. When I try dailing a local or long distance number I get an "All circuits are busy now" message from the SIP client. There is a message at the top of the SIP/bv trunk page in AMP mentioning that this trunk isn't included in any routes which I assume is the problem. Is this related to the fact that I used Asterisk@Home v1.3 or is there some missing step? [WM: Sorry. There was a missing step. Kinda reminds me of my grandmother always leaving one ingredient out of the recipes she handed out to make sure hers always tasted a little bit better. Look at the article again and complete the step labeled: Configuring Asterisk for Outgoing Calls. My apologies.]