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  #11 (permalink)  
Old January 26th, 2005, 04:53 PM
chucken chucken is offline
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Phoneboy,

This is a great script. Is there any chance of getting a copy of the underlying php that does the asterisk configuration? I've been setting a bunch of these up for family and friends, and I'd like to set up something custom, that's as close to a 1-click config as I can get (don't want grandma to have to understand dial plans, etc.)

I packet traced a POST of the form, and I can guess what a lot of the value-attribute pairs mean, but others don't mean much without some kind of Sipura guide. Is there such a document that end-users like myself can get a hold of?
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  #12 (permalink)  
Old January 26th, 2005, 10:56 PM
ichilton ichilton is offline
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Quote:
Originally Posted by chucken
Is there any chance of getting a copy of the underlying php that does the asterisk configuration?
Yeah, I too would be interested in this!

Thanks

--ian
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  #13 (permalink)  
Old January 27th, 2005, 06:33 AM
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PhoneBoy PhoneBoy is offline
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The backend is actually a perl script.

The name/value pairs were reverse engineered from the web interface of the SPA3000. The answers you seek are in the source of the web interface.
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  #14 (permalink)  
Old February 20th, 2005, 03:16 AM
_Belial _Belial is offline
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Not sure if it's me or perhaps the latest firmware (2.0.13) but the wizard creates a config which doesnt' work for me, no FXO calls are coming through to asterisk at all. I pasted the extensions.conf and sip.conf that it generated, reloaded aterisk but no joy

sip debug shows nothing at all
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  #15 (permalink)  
Old February 20th, 2005, 07:44 PM
mike1267 mike1267 is offline
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Great Script,

Works for me, but I have a couple of questions:

Is is poiisble for the SPA3K to alter asterisk of an inomming call without answering it? I would like to decide if to answer the call at the moment the incomming call does not start ringing on asterisk untill its answered by the SPA and the caller then hears an inband ringing tone.

Also, I too am having problems making outgoing calls, I suspect bacuase the pstn=spa3k has not registered and therefore asterisk does not know where to send the invite?
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Old February 20th, 2005, 07:44 PM
  #16 (permalink)  
Old February 21st, 2005, 12:52 AM
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mberlant mberlant is offline
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mike1267,

It only takes a few moments to read what others have posted before asking a question that has been addressed many times before. There is a wonderful discussion about Asterisk handling Caller ID prior to allowing the SPA3000 to accept the incoming call. It is here.

As for handling outgoing calls, how do you know that the SPA's PSTN Line has not registered? Does it say this on the Info screen?
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  #17 (permalink)  
Old March 25th, 2005, 05:53 PM
MillsapsPE MillsapsPE is offline
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I have my SPA3000 working (by using a different set of instructions). By the looks of the wizard page, it appears as if the wizard will do the same thing I already have (but I need to actually try it to confirm).

My question is, once you use the wizard to set up the SPA3000, will incoming CID info be transferred through the SPA3000 to Asterisk? With my current setup, Asterisk is sending the Extension Name and Extension Number as the callerid info to my SPA841. If I set the PSTN CID for VoIP CID to yes, * stops working (call rings twice, as before, then just silence).
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  #18 (permalink)  
Old May 10th, 2005, 06:27 PM
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I have updated the SPA3K Asterisk Wizard. I've made some minor tweaks to the Sipura configuration and also added a few lines in the same Asterisk config that should help this work a lot better. Comments welcome.
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  #19 (permalink)  
Old August 2nd, 2005, 12:10 PM
NachoR NachoR is offline
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Default asterisk@home --> SIPURA3000 (wiht no Off Hook While Call

Hello,

I use the wizard to configure my Sipura 3000 and works great, but when I call from some extencion to the PSTN using the sipura, I get one ring.. and after a two second silence I get a "funny" ring tone (this is the sipura dialing thrue the PSTN line). I think the new firmware version 3.1.5(GWb) will be solve the problem, but no. I find the option "Off Hook While Calling VoIP" in the PSTN-To-VoIP Gateway, but can
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  #20 (permalink)  
Old August 11th, 2005, 12:03 PM
DaveQuinn DaveQuinn is offline
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Hi

I am pretty clueless so please bear with me.

I setup an astersik system so it works internally as a home pabx. This works well with voice-mail etc. Have not been able to get to the outside world via IP providers, and no link to PSTN. Not particularly bothered about VoIP at this point in time, but would like to go from home phones and softphones to PSTN via sipura 3000.

I tried the wizard and I haven't got a clue whats happening.

Any guides for the absolute dummy anywhere?
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Old August 11th, 2005, 12:03 PM
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