[shahzadmasih]

Hi, This post is very informative, however I would like some specific information. If someone can help me then please send me a private message. Best Regards,

[eric]

Please post your question to the forum.

[cristian_paun]

One of my question was if exist any wizard to config a spa3000 as extension of asterisk. Spa3000 is not in same lan with asterisk.

[Zvonimir Delač]

Originally Posted by PhoneBoy

Now available for your testing pleasure:

Voxilla - Linksys SPA 3xxx Configuration Wizard for Asterisk

Might also work with other PBX-type systems as well. It does generate some basic sample configs for Asterisk. Feedback welcome, of course.

I'll tried several times to run wizard but failed, becouse this link is unrecheable. Can someboy adviced how to use this wizard?

Thanks,

Zvonko

[eric]

The link should work. What error message do you receive?

[packetvoice]

I just ran the Voxilla SPA3000 Configuration Wizard for Asterisk. The wizard came up with the setting for the PSTN line as follows:

Register=No

Is this the correct setting for incoming PSTN calls to ring the phone connected to "phone" port of the SPA3000?

The reason I ask this question is becuase sevaral posts in this thread "complained" that their PSTN line was NOT registering. The question is:

Does the PSTN line need to register for the wizard settings to work properly?

Thanks.

[larryalk]

I also have the problem reported so many times on this thread where the Sipura pstn line is 'not registered' and an incoming pstn call rings once and is seemingly 'swallowed up' by Asterisk.

My system is nearly ready to deploy but the one thing missing is the ability to receive incoming pots calls over the pstn line and send them to Asterisk so they can be picked up either by pots phones connected to Sipura or sip phones. I've been working for several days on this problem. I see in debug where the 'invite' fails to the 'line 8' extension which is only a skeleton right now.

Unfortunately the 'line 8' dial plan is not well explained. There is no sample sip.conf extension for the extension I specified in Wizard Step 10 and no corresponding extensions.conf context in the Wizard printout.

I'm pretty sure if there were proper entries above then the Sipura pstn line would register and accept the call.

I humbly and respectfully request that a sample sip.conf extension and it's corresponding extensions.conf context be posted here. I need to finish up and deploy the VoIP system, as I promised my wife, before she returns from a family trip in two weeks or my name is mud.

Larryalk

[larryalk]

Where can I find the perl? source code for the SPA3000 wizard?

[eric]

The wizard source code is not publicly available.

[larryalk]

Thanks for your prompt reply Eric.

I have used the SPA3000 configuration wizard but am unable to receive POTS line and intercom calls to POTS phones connected to the SPA and wanted to view the perl code to better understand some the function.

Is there a more detailed explanation than my printout of the steps and my answers?

Specifically, what is the funtion of the "asterisk" extension for "VoIP to PSTN calls" specified in step 9? Does it use Dial Plan 8? What calls it?

What would be a more descriptive name than "spa3k-pstn-out" which I currently use?

BTW, I am told there seems to be a bug that specified Dial Plan 8 as (<S0:66618>) instead of (S0<:66618>). Or perhaps it can be specified either way?

Larry