[WM: Sure. That functionality is now built into the new inbound trunks setup in freePBX. Just use one of our preconfigured TrixBox systems at the top of the page to get started.]

1 cent per minute to all of USA and Canada!

Take a look at GenericVoIP.net they are now making some changes and will soon be offering more destinations on the 1 cent per minute calling. They do now also support caller ID contrary to what the web site still says. It is truly a good deal.

[WM: Heh, heh. OK, I’ll post your comment but not without a few of my own. First, if you thought TelaSIP has a lousy web site, you’ll love this one. Second, if you want CallerID, that ups the price to 1.7¢ per minute. Then there’s the little matter of rounding calls to the whole minute. Gee, what a deal! Finally, there’s the fine print (and more) below. So… I think I’ll pass.]

*Price quoted is cash price. PayPal price is 4% more. Credit Card is 5.5% more. USA 48 at 1 Cent Per minute requires a minimum of 5000 minutes per month, funded with a single payment of $50.00 per month. A minimum usage of $50.00 per calendar month applies to all $0.01 per minute accounts. All calls billed in 60/60 Intervals. USA 1 Cent accounts do not allow international calls. Use of PayPal requires a VERIFIED PayPal account. USA 48 / Canada $0.017 accounts with no outbound traffic for 60 days are subject to deletion.

[WM: This article was written a long time ago when I, too, was a "newbie." I’ll have another look at it in the "modern" context of TrixBox. Thanks for the suggestion.]

I’m dialling out using PRI. After about 10Sec I’m getting busy tone when the remote side didn’t answer.
Is this value can be reconfigured??.

TIA,

"I have one problem with auto attendent – it all works, and the volume is great from any extensions, but incoming callers ona pstn (X100P) card hear the recording at very low volume. Any suggestions?"

How would I go about raising the volume on the auto attendent? Any suggestions would be greatly appreciated.

Thanks!

-Adrian

[WM: Solved in TrixBox.]

exten => 111,8,GotoIf($["${CALLERIDNUM:0:7}" = "Private"]?who-r-u,s,1)
exten => 111,9,GotoIf($["${CALLERIDNAME:0:7}" = "Private"]?who-r-u,s,1)

exten => 111,12,GotoIf($["${CALLERIDNUM:0:4}" = "PSTN"]?who-r-u,s,1)
exten => 111,13,GotoIf($["${CALLERIDNAME:0:4}" = "PSTN"]?who-r-u,s,1)

[WM: Some providers sent Private as the number and others send it as the name. Same with PSTN.]

For the benefit of newbies like me, can you please provide the exten=>99 code to support the SPA-3000?

[WM: Someone did it for me. Here’s the link.]

[WM: You’ll need to buy a Sipura SPA-3000. Then just follow our tutorial to get it going, and you can do exactly what you suggest.]

exten => s,1,Dial(SIP/6781234567@bv,60,r) should it be this instead:
exten => s,1,Dial(ZAP/6781234567@1,60,r)

Also, after my main message is played it takes a while to ring to my default extension which is 201. i have the t option set.

My main message is like: "You have called Kim’s asterisk PBX system (Want to show off others ;)) please wait a moment while you’re call is being connected. If it is an urgent matter you can dial 5 to connect to my work phone"

Then I have 2 other hidden options: one for DISA and other is the ‘t’ option which is basically going to my default ext 201. And for 201 I have voicemail. So what I want is to minimize the wait time before I can hear ring back tones calling my 201 extension. Also there is a long silect between. Can you help….

I am trying to setup an autoattendant that dials my cellphone when someone calls my home and I’m not there. The problem is in waiting 10/20/30 seconds for me to pick up my cellphone before returning to my asterisk voicemail. If for example, my cellphone is off, my cellphone voicemail will pickup immediately, and the caller will not get a chance to leave a voicemail message on the asterisk machine, but rather on my cellphone.

Is there a way to have me autheticate and/or accept the incoming call on my cellphone, or else the call is transfered back to asterisk voicemail? Essentially, this way it will never seem to an outside caller that the call was transfered to my cellphone (i.e. they will never get my cellphone voicemail).

Thanks a ton, keep up the awesome discoveries.

Chris

You indicate:
"You do need to modify the [ext-park] context code (line 3) to specify the extension, extensions, or ring group where you want this incoming call announced. We use 222 as a ring group for our entire house."

I use AMP to create a Ring Group, which created Ring Group 1. It has 5 extensions in it. So I tried the following:
exten => 70,3,ParkAndAnnounce(/tmp/asterisk-stranger:vm-isonphone:at-following-number:PARKED|40|local/1@from-internal|who-r-u,s,7)

But that doesn’t work. I am not sure what to change "local/200@from-internal" to to reference a ring group instead of an extension.

Any suggestions?

Thanks and I will keep reading these great tutorials.

Dave

[WM: I wouldn’t use 1 for a ring group. Think of that number as the extension you dial to ring all the phones in the ring group. That’s why we use 222. Normally, the syntax for ringing a ring group is Dial(local/222@from-internal,18,m) where 222 is the ring group, 18 is the time to ring the extensions, and m means play music on hold while ringing. This may not work with Call Parking but it works fine in your normal dial plan.]

[WM: It depends upon the provider. This range covers most of them that we’ve encountered.]

[WM: Bill, If I had to dig into AMP that deeply, I’d be tempted to rewrite it. But, at least today, I’m not that tempted. Like a lot of developers, the AMP folks appear to have intentionally made a lot of the code obscure to preserve their turf. But who knows maybe these guys are so smart that they actually like drawing pictures based upon reflections in the mirror.]

I wonder if that is do-able?

[WM: Absolutely. We prefer MySQL for the task since storing data in ast_db is sorta like putting your address book in the Windows Registry, but each to his or her own. Here’s a link that will give you some ideas and get you started.]

i have setup my asterisk@home+spa300 using your great guide. Anyway i have a stange propblem with digital recepcionist.

I’m using:

exten => 298,1,GotoIf($["${CALLERIDNUM:0:2}" = "00?]?2:3)
exten => 298,2,SetCIDNum(${CALLERIDNUM:2})
exten => 298,3,SetMusicOnHold(default)
exten => 298,4,Goto(aa_1,s,1)

Thats fine..i set digital recep. with several options and so..the problem is:

– when caller push 1 it gos to extension sip softphone 201 , thats working fine

– When caller push 2 it goes to extension sip line1 sipura , so cordless phone should ring, but it doesnt work fine, it repeats 1st recepcionist message aa_1 .

When i use simply pstn->sip extension incoming calls, i can set cordless ext. fine, it rings fine and i can talk fine.

So whats wrong?

[WM: Look through your [from-internal-custom] context in extensions_custom.conf for all the entries that start with exten=>2,… All of those get processed every time there is an incoming call. Be sure you don’t have one in there that you didn’t remember.]

If you can help that would be great.

Chris

[WM: Your best bet is one of the forums we’ve recommended in our columns: SourceForge or Voxilla.]

Here is a portion of my call output
——————————————–
dialparties.agi: Dial string is Local/111@from-internal|30|tr
— AGI Script dialparties.agi completed, returning 0
— Executing Dial("Zap/1-1″, "Local/111@from-internal|30|tr") in new stack
— Executing Wait("Local/111@from-internal-a0e8,2″, "1″) in new stack
— Called 111@from-internal
— Executing SetMusicOnHold("Local/111@from-internal-a0e8,2″, "default") in new stack
— Executing GotoIf("Local/111@from-internal-a0e8,2″, ""?｢ﾀﾜ?ｰ?ﾉて??ﾇｬｨ?ﾇｬ｢"?who-r-u|s|1″) in new stack
— Executing GotoIf("Local/111@from-internal-a0e8,2″, ""foo?｢ﾀﾜ?ｰ?ﾉて??ﾇｬｨ?ﾇｬ｢foo"?who-r-u|s|1″) in new stack
— Executing GotoIf("Local/111@from-internal-a0e8,2″, ""?｢ﾀﾜ?ｰ?ﾉて??ﾇｬｨ?ﾇｬ｢Anonymous"?who-r-u|s|1″) in new stack
— Executing GotoIf("Local/111@from-internal-a0e8,2″, "0?who-r-u|s|1″) in new stack
— Executing GotoIf("Local/111@from-internal-a0e8,2″, "0?who-r-u|s|1″) in new stack
— Executing GotoIf("Local/111@from-internal-a0e8,2″, ""?｢ﾀﾜ?ｰ?ﾉて??ﾇｬｨ?ﾇｬ｢Private"?who-r-u|s|1″) in new stack
— Executing GotoIf("Local/111@from-internal-a0e8,2″, "0?who-r-u|s|1″) in new stack
— Executing GotoIf("Local/111@from-internal-a0e8,2″, ""?｢ﾀﾜ?ｰ?ﾉて??ﾇｬｨ?ﾇｬ｢OUT OF AREA"?who-r-u|s|1″) in new stack
— Executing GotoIf("Local/111@from-internal-a0e8,2″, "0?who-r-u|s|1″) in new stack
— Executing GotoIf("Local/111@from-internal-a0e8,2″, "0?who-r-u|s|1″) in new stack
— Executing DigitTimeout("Local/111@from-internal-a0e8,2″, "3″) in new stack
— Set Digit Timeout to 3
— Executing ResponseTimeout("Local/111@from-internal-a0e8,2″, "3″) in new stack
— Set Response Timeout to 3
— Executing BackGround("Local/111@from-internal-a0e8,2″, "custom/aa_1″) in new stack
— Local/111@from-internal-a0e8,1 answered Zap/1-1
— Playing ‘custom/aa_1’ (language ‘en’)
———————————————————–
How can I fix this?

[WM: Replace the typographic quotation marks in the script with regular quotation marks.]

Thanks
That_Kid

[WM: Check carefully for typos.]