Taken the Astrisk Plung, its DEEP... need help :(

rizsher · #1 (**permalink**) July 8th, 2006, 03:49 AM

Hi,

I finalyl installed Tribox on an old laptop... took a few hours, but after following the excellent how-to at NerdVittles, I managed to load and get astrisk working to a certain point, I could get my SPA3102 to register to asterik, outgoing calls were mixed.. some would go thru, but mostly I would get a msg saying "All Circuis are busy", which I know isn;t a message from my VSP.

Anyways, in trying to fix this issue, I followed many many tutorials avaialble on the net, and now NOTHING works

COuld somes please tell me:

1) How to initialise Asterisk so I get a clean asterisk config without actually goign thru the entire re-install/update routine

2) Does anyone know of a tutorial for interfacing SPAXxxx with Asterisk?..

Thanks.
Rizwan

mberlant · #2 (**permalink**) July 8th, 2006, 04:17 AM

I don't think there is an easier way to perform a "factory reset" on Trixbox than to do a complete reinstall.

As to configuring SPAs to work with Asterisk, I have done a couple of hundred by using the Voxilla Wizard for "FWD w/STUN" and then replacing fwd.pulver.com with the Asterisk's FQDN once the configuration is in the SPA. On the Asterisk side I configure by hand, so I can't comment on what Trixbox does except to say that Asterisk itself is very forgiving and, so, whatever Trixbox does within sip.conf should allow your SPA to connect successfully.

rizsher · #3 (**permalink**) July 8th, 2006, 10:55 PM

Thanks Michael.

I did a reinstall and have now managed to achieve the following:

1) My Voiptalk.org account is registered in Asterisk
2) My SPA3102 (connected to the router) is now pointed to the Asterisk server and I can now make and receive calls thru VoIPTalk through it.
3) I can call my PSTN line number from my cellphone, get authenticated by CallerID, and make voIP Calls thru VoIPTalk registered under Asterisk.

What I would now like to do is enable VoIP to PSTN gateway such that, when a call comes to my VoIPTalk Incoming DID, it should ring a few times, after that, callers to the VoIP number should be present with a voice prompt to the effect "We're not home rihgt now, please press 1 for xyz, 2 for abc and 3 for def". This in turn, should forward the call thru the SPA's PSTN-linae to the xyz/abc or def's callphone, lastly, if if no key is pressed, to forward to xyz's cellphone. Before the "plunge", I had the SPA3102 such that any call would get forwarded to MY cellphone after 4 rings, this left the mrs out and had me on the receiving end of a lot of calls from the in-laws not really meant for me!!!!.

Is this feasible... if so, how?.

One more thing mberlant, I remeber reading up one of the posts from you sometime ago, where you mentioned somethign about reloading a file whenever the dyndns IP address is updated, and you had a way of automating the process. Can you plz point me to it?.

Thanks for all your help.

rizsher · #4 (**permalink**) July 8th, 2006, 11:02 PM

Hmmm...

Something isn't quite right. I was doing my testing using another VoIPTalk account configured on a Softphone on my desktop, and making Sip to PSTN number calls (not SIP to SIP extensions) to the voiptalk account on Asterisk, and they were ringing the account just fine... however, when I try dialling the same number from my Egyptian hecellphone, immediately get a recording saying "Please hold while we try to connect you"....repeatedly...

This is a the cal log. Calls 0 and 1 are fron the SoftPhone.. all the rest are from my cellphone...the only difference I can see is, in the softphone account, there is "unknown" in the CLID, where as the cellphen CLID is like "0020106074366" <0020106074

Code:

 Calldate    Channel  Source  Clid  Dst  Disposition  Duration  

0.  2006-07-08 16:16:04 SIP/811381... 2131665134 "unknown" <2131665134> 204 NO ANSWER 00:02 
1.  2006-07-08 16:13:02 SIP/811381... 2131665134 "unknown" <2131665134> 204 NO ANSWER 00:03 
2.  2006-07-08 16:11:41 Local/99@f... 0020105555555 "0020105555555" <0020105555 t ANSWERED 00:03 
3.  2006-07-08 16:11:34 Local/99@f... 0020105555555 "0020105555555" <0020105555 t ANSWERED 00:10 
4.  2006-07-08 16:11:28 Local/99@f... 0020105555555 "0020105555555" <0020105555 t ANSWERED 00:10 
5.  2006-07-08 16:11:27 SIP/811381... 0020105555555 "0020105555555" <0020105555 204 ANSWERED 00:17 
6.  2006-07-08 16:08:49 Local/99@f... 0020105555555 "0020105555555" <0020105555 99 ANSWERED 00:01 
7.  2006-07-08 16:08:42 Local/99@f... 0020105555555 "0020105555555" <0020105555 t ANSWERED 00:08 
8.  2006-07-08 16:08:42 SIP/811381... 0020105555555 "0020105555555" <0020105555 204 ANSWERED 00:08 
9.  2006-07-08 16:04:23 Local/99@f... 0020105555555 "0020105555555" <0020105555 99 ANSWERED 00:01 
10.  2006-07-08 16:04:17 Local/99@f... 0020105555555 "0020105555555" <0020105555 t ANSWERED 00:07 
11.  2006-07-08 16:04:10 Local/99@f... 0020105555555 "0020105555555" <0020105555 t ANSWERED 00:11

mberlant · #5 (**permalink**) July 9th, 2006, 03:09 AM

How did you integrate your 3102 with Asterisk? Did you use the Voxilla Wizard for SPA-3000/Asterisk? If you do, it sets up PSTN Line with HTTP Digest as the authorization method, which facilitates one-stage in and out calling.

Another trick to use in this integration is to assign PSTN Line an extension number that looks like a Trunk Group Access Code. So, if you have assigned VoIPTalk "9" as its Access Code you might assign PSTN Line "Extension 8".

The other thing you mentioned is IVR. The easiest way to implement an IVR is to write the script and assign an extension number to the entry point of the script. Here's some sample code for you to modify:

Code:

[default]

exten =>555,1,Goto(ivr-main,s,1)

; ----------------------------------------------------------------------------
; IVR processing
;
[ivr-main] ; For use when SPA-3k answers ringing POTS line.
;
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,NoOp(${CALLERID(num)})
exten => s,4,NoOp(${CALLERID(name)})
exten => s,5,DigitTimeout,5
exten => s,6,ResponseTimeout,10
exten => s,7,Playback(jp/vm-nobodyavail)
exten => s,8,Playback(jp/vm-enter-num-to-call)
exten => s,9,Background(jp/vm-Friends)
exten => s,10,Background(jp/jp-wa)
exten => s,11,Background(digits/jp/5)
exten => s,12,Background(jp/jp-wo)
exten => s,13,Background(jp/jp-oshitekudasai)
exten => s,14,Background(jp/privacy-thankyou)
exten => s,15,Background(silence/10)
exten => s,16,Background(silence/10)
exten => s,17,Hangup()
;
exten => _1XXX,1,Goto(default,${EXTEN},1)
;
exten => 2,1,Playback(that-tickles)
exten => 2,2,Wait(1)
exten => 2,3,Goto(s,26)
;
exten => 3,1,Playback(carried-away-by-monkeys)
exten => 3,2,Wait(1)
exten => 3,3,Goto(s,26)
;
exten => 4,1,Playback(weasels-eaten-phonesys)
exten => 4,2,Wait(1)
exten => 4,3,Goto(s,26)
;
exten => 5,1,Playback(wait-offensive-sounds)
exten => 5,2,Wait(1)
exten => 5,3,Goto(s,26)
;
exten => 6,1,Playback(why-no-answer-mystery)
exten => 6,2,Wait(1)
exten => 6,3,Goto(s,26)
;
exten => 7,1,Playback(jedi-extension-trick)
exten => 7,2,Wait(1)
exten => 7,3,Goto(s,26)
;
exten => 8,1,Playback(away-naughty-boy)
exten => 8,2,Wait(1)
exten => 8,3,Goto(s,26)
;
exten => 9,1,Playback(wrong-try-again-smarty)
exten => 9,2,Wait(1)
exten => 9,3,Goto(s,26)
;
exten => 0,1,GotoIfTime(${MB_SLEEP}|*|*|*?102)
exten => 0,2,Goto(find-michael,s,1)
exten => 0,102,Playback(what-time-it-is)
exten => 0,103,Background(vm-leavemsg)
exten => 0,104,Background(or)
exten => 0,105,Playback(pls-hold-while-try)
exten => 0,106,Background(silence/5)
exten => 0,107,Goto(find-michael,s,1)
;
; -----------------------------------------------------------------------------

And, let's see if your inbound PSTN problem doesn't clear itself up with the HTTP Auth change (assuming you're not already there).

rizsher · #6 (**permalink**) July 9th, 2006, 06:30 AM

Thanks again Mberlant.

Off to work just now, will try to follow your instructions when i get back. The script goes under sip.conf?.

Just about ur last comment, the problem is not with inbound PSTN, infact, I don't use my home PSTN line to receive OR make calls...its almost exclusively used for PSTN-VoIP and VoIP-PSTN bridging. The problem I explained is with my VoIPTalk account which used to be configured on Line 1 of the SPA3102 before i switched to Asterisk, and now, calls from a softphone, still to my VoIPTalk DID, so its not a SIP call, work, but calling the same DID number from my cellphone gives the recording.

Thanks.
Rizwan