Comments on: Turbocharging Your Asterisk@Home PBX

By: Villalvilla

Villalvilla — Mon, 27 Jul 2009 19:52:50 +0000

Nice article! But... You have an error. When you configure PSTN for outbound calls, you need to configure the "PSTN user" tab. Just put "00*" in "Cfwd Sel1 Caller" and "199" in "Cfwd Sel1 Dest", instead of putting that information in the "user 1" tab. The rest of the article works like a charm! congratulations.

By: Fred

Fred — Tue, 17 Jul 2007 13:45:48 +0000

Nice Tutorial thanks. I’m trying to follow along but I installed AsteriskNOW which seems very different. The "trunks" part doesn’t mak much sense so far.
Whay are there so many variants of asterix, is @home the best one for a beginner? I thought NOW was supposed to be?
Thx for any guidence.

[WM: This is a very old article. AsteriskNOW is different, and Asterisk@Home has been superseded. Click on one of the links at the top of our web page to download a turnkey solution for Linux, Windows, or a Mac.]

By: Ian Worthington

Ian Worthington — Mon, 19 Feb 2007 15:16:22 +0000

Hi Ward.

10 months after I bought my spa-3000, swmbo is finally out of town giving me a chance to integrate this wee timorous beastie with my trixbox (in your vm box).

I had some problems with your instructions, and some reservations about defining it as an extension rather than a trunk, and Kerry Garrison’s instructions didn’t work either.

But I wrestled it to the ground and all’s now well. I also found out what needed changing for it to work with a UK BT circuit. For anyone having similar issues I’ve documented it all at http://isw.me.uk/content/view/14/28/

ian
…

[WM: Many thanks. Our intstructions are getting a little long in the tooth and, unfortunately, we no longer have a PSTN line to play with. We’re a pure IP shop and home at this point.]

By: Dave

Dave — Fri, 05 Jan 2007 17:57:26 +0000

The SPA3000 setup here is a hack because it makes the incoming calls appear to come from an extension on the network and not from a trunk line. This may give you problems with DID and IVR tone recognition. Here is a better set of instructions. [WM: To each his own. Thanks for the link. We've never had a problem with DID's or IVR tone recognition.]

By: Duncan

Duncan — Sun, 31 Dec 2006 01:39:39 +0000

Thanks for this great info. I have Nerdvittles V1.2.3 trixbox running in VMWare Player on an Athlon XP2800+ with 1GB RAM and mirrored SATA drives, a Linksys SPA-3000 configured as detailed in this how-to, and a Polycom SoundPoint IP 501 SIP. Thanks to your instructions, I can dial in and out using my PSTN line via the SPA-3000. Problem is, the sound cuts in and out so badly it is unusable. Sometime I get no sound for 5-10 seconds at a time. Is this just the VMWare implementation? Is there something else I should check? Is this a possible result of improper configuration? I configured my Polycom SIP phone by trial and error so I certainly could have done something wrong there. any idea where to look? Is dedicating a server the only way to go? Thanks.

By: Josh

Josh — Mon, 20 Nov 2006 01:59:16 +0000

Just thought i'd share my experience with the SPA3102. I used the guide here to set up Trixbox and the SPA3102. Also if you're not using the 3102 as a router see my post here. I'm also curious as to the differences between the Voip-info setup (which I used) and the setup described here. Calling in to my PSTN rings all my extensions (ring group) and calling out calls out normally. I can dial 101 to get my extension on the 3102 and from there I can dial 200/201 to get my softphones. If you need to reach me, do so through voxilla.

By: Jens

Jens — Tue, 17 Oct 2006 23:44:50 +0000

Great guide. However I have one problem. It appears that my SPA-3102 doesn’t forward any DTMF sounds to Asterisk. Is there an additional setting I am missing?

Thanks,
Jens

By: tom glaab

tom glaab — Sat, 23 Sep 2006 19:10:56 +0000

Alfred — tak a look at #37. That solution gets rid of the whole extension 99/199 schema and seems to be working well with my Trixbox and SPA-3000.

By: Alfred

Alfred — Fri, 22 Sep 2006 05:41:34 +0000

If I send the call from the SPA3102 to 7777 caller ID works. Everything works. What do I need to add to my dial plan to pass the Caller ID info when transfering to extension 99?

I need to send to 99 as the phone is not answered until the 4th ring when sending to 7777. When I send to 99 I have it setup to pickup on the first ring. However, the name I have setup in the SPA3102 comes across as the caller ID info.

Help!

By: Alfred

Alfred — Mon, 18 Sep 2006 06:21:59 +0000

New question. If I want the PSTN line which is setup as a trunk to work the same as the VoIP inbound calls, what do I do? The above instructions discuss ringing a phone extension. I want the PSTN line to follow the logic setup in Inbound Routes. Currently this is set to Time Conditions.

Everything is working as currently the call is sent to extension 99 but then does not know what to do since I did not paste the code into Extensions_Custom.

I am using the latest version of Trixbox with VM Ware.

By: Alfred

Alfred — Mon, 18 Sep 2006 06:17:12 +0000

Solved my own problem. FYI: After connecting to the SPA-3102 via the Ethernet port, click on WAN and turn on Web Access. Then disconnect from the Ethernet Port and plug into the WAN port. All of the stuff only works via the WAN port. This assumes that you already have a router and are not using the SPA-3102 as a router where you would use both the Ethernet and WAN jacks.

By: Alfred

Alfred — Sun, 17 Sep 2006 23:41:25 +0000

After setup my Line 1 Status shows registered but the PSTN Line Status states Failed. The User Name 204 in PSTN Line will then change the PSTN Line Status to registered. Trying 99, 199, asterisk and the phone # which is the name of the Trunk all fail. Please share thoughts.

I am using a TrixBox and SPA3102.

[WM: SPA-3102 is a very different beast than the 3000, and we have not yet tested one of them. Sorry. Visit the Voxilla Forums.]

By: Tom

Tom — Sat, 16 Sep 2006 13:59:08 +0000

Regarding #32 above, I think I found it. See: http://www.aussievoip.com/wiki/index.php?page_id=119&tk=d2dbfa84507b907bcb25&refresh=1. At first look things are working in both directions. While firmware v3 loaded on my Sipura v2 hardware, I went back to v2 FW just to eliminate that variable. Finally my wife won’t constantly complain about the phones (I hope 🙂

By: Andy Nilssen

Andy Nilssen — Sun, 10 Sep 2006 18:31:14 +0000

If you’re pulling your hair out trying to make an outbound PSTN call with the SPA3000 (i was), here’s one thing to check: Line-In-Use Voltage. Turns out my PSTN line is a low voltage line and the SPA3000 was not recognizing it. The fix is easy. See the only Point #10 on this page: http://www.nch.com.au/hardware/setup/fxo.html

By: tphank

tphank — Wed, 30 Aug 2006 22:14:39 +0000

I’ve been using Asterisk since 2.7, now 2.8 and any day now will migrate to Trixbox… Now at home, I have an SPA3000 setup as an extension (100). When a specific stanaphone number is called, I use a DIAL(SIP/100). It appears that asterisk or the SPA has answered the call and is ringing while the extension is being called. Is there a way for the call not to be answered? I know it is being answered, since when I call by cell, by the first ring my cell shows connected, even though the extension is just starting to ring…. I Just have NoOp commands before the dial, so it must either be the dial or the SPA answering the call? Is there anyway to not have the call answered unless someone picks up the extension?
Thanks!
tphank

By: Jens

Jens — Tue, 29 Aug 2006 17:24:31 +0000

This is a great tutorial, but I have one question: What does do? Is the 99 a link to extension 99?

Thanks,
Jens

By: Vincent

Vincent — Fri, 18 Aug 2006 23:34:14 +0000

I'm using the Linksys 3102, the newer version of the 3000, and the Linksys will forward incoming PSTN calls to the PBX by just setting an extension in User1 > Call Forward Settings > Cfwd All Dest. Using a dial plan in "PSTN Line" doesn't do anything, and prepending numbers to the caller ID number isn't needed. Unfortunately, no one has yet been able to solve a serious issue: When forwarding calls to the PBX (I guess it's really FXS > FXO > PBX, the Linksys goes off-hook while notifying the PBX, so even if no one ends up answering the call, the caller will be charged for the call. I've tried a bunch of settings in the Linksys interface, including the "Off Hook While Calling VoIP = no (default anyway)" in PSTN Line, all to no avail. If someone knows how to disable this, I'm interested. [WM: I'd recommend you repost this on the Voxilla forums. There are lots of Linksys/Sipura experts there.]

By: tom glaab

tom glaab — Sat, 29 Jul 2006 19:25:45 +0000

Since upgrading to Trixbox 1.1 I have not received caller ID from my SPA-3000. The SPA decodes it OK, but it doesn’t seem to make it any further (log analysis to follow). A post at http://forums.whirlpool.net.au/forum-replies-archive.cfm/549271.html suggests that Asterisk and Trixbox interact differently with the SPA, meaning this tutorial won’t work with Trixbox.

My SPA-3000 also answers on one ring, no matter what PSTN Answer Delay is set to. Arrgh. Can I load v3 firmware on a box with v2 firmware/hardware?

tg.

By: martin

martin — Fri, 28 Jul 2006 04:09:13 +0000

How do you point the sipura 3000 to the attendant ?

By: Rizwan

Rizwan — Tue, 11 Jul 2006 09:47:30 +0000

"be sure to replace the WordPress-inserted, front and back quotes with normal quotation marks, or you?��Ǩ�Ѣll send Asterisk into the ozone."

Could you please explain what you mean by "normal quotation" marks? Is it ‘ ‘ or ` ` ??.

Thanks.
Rizwan

[WM: It’s quotation marks where the one on the left of a quotation looks different than the one on the right, i.e. not the one two doors down from the L key on your keyboard.]

By: Brian

Brian — Fri, 16 Jun 2006 22:06:01 +0000

I have an old Linksys RTP-300 from my Vonage days that now has the 3.1.10 NA firmware on it. Any tips to program the 2 fxs ports on it? Thanks for all of you help with AAH and now TrixBox.

By: joshua

joshua — Mon, 12 Jun 2006 01:28:36 +0000

MOH w/Asterisk and hte SPA-3k —- please explain how to..

By: Olu Solaru

Olu Solaru — Sat, 13 May 2006 06:46:02 +0000

Great instructions. unfortunately I have a Sipura 1001. I have been unable to get any similar information for my Sipura 1001, every information I have researched has been very sketchy. I guess some of your instructions don’t apply to the Sipura 1001. But if you know where I can get some REAL information. I will really appreciate it.. Thanks..

By: Glenn Jensen

Glenn Jensen — Mon, 17 Apr 2006 03:06:34 +0000

Thanks – I can make calls from a phone attached to the Sipura, but I get the dreaded ‘all circuits busy’ error calling out the Sipura trunk. Pulling my hair out….. all ideas welcome.