[gatorback]

Asterisk 1.8 is running on an ASUS RT-N16 (Optware) attempting to configure Google Voice. In this thread, I am interested in validating the connection between a SPA2102 ATA and *1.8


I created a SIP extension for SPA2102 ATA based on this article.

[okki]

I may be wrong, but you may not understand the correct idea behind such a setup. The way to set up Asterisk for regular PBX use is usually:

IP-phones ==> Asterisk on router ==> Internet

meaning your IP phones (or ATA) register to Asterisk, which then processes the call based on the dialplan you provided in extensions.conf and/or extensions.ael.

In particular, understand the NAT settings. E.g. if you don't use the directmedia=yes option (the default) you also don't need the nat=yes option in the SIP entry for the SPA.

If you understand all that, I suggest you read up about Asterisk on Optware http://www.dd-wrt.com/phpBB2/viewtopic.php?t=43787. Read all those pages (only 25 ); it has a lot of useful information, like which modules to load and also about Google Voice.

If you think the documentation is very bad, I concur. I intended to make a nice wiki page on Asterisk, but I have to find the time. Also, the fact that apparently I cannot register an account to the wiki site without a hassle is holding me back. (Hint to the site owners .)

Success!

[gatorback]

The objective of this posting is to verify that connection:

IP-phones <==> Asterisk on router

is properly configured. Verification implies that I would have a test: preferable from the CLI.

For example, when I add a network device, I usually use:

[okki]

If that's the only thing you want to know, well, yes - your SPA does communicate Asterisk, otherwise it wouldn't recognise it at the 192.168.8.110 address.

So the next step is configuring Asterisk, for which I advise the above in my previous post. (And other people's advice in your other thread.)

[gatorback]

@okki: Thanks for the feedback. Before I move on to configuring jabber.conf and extensions. conf, Is there a test that can be performed that would involved dialing the ATA's phone and receiving some audible response from Asterisk?

I would feel better that a validation test was performed. Verification (above) is good, but validation would be stronger and provide higher confidence.

My goal is to confirm that the configuration between the ATA & Asterisk is complete and correct.

[okki]

Sure, but Asterisk is so flexible, there's no standard test or something. And understand there are different configurations for different purposes, which are all correct.

Still, getting an audio response is a good test to see if it works. Though it would involve entering something in your dialplan. Asterisk has some nice audio files (be sure to download them via opt-ipkg). You can call them for example with

exten => 1234,1,Playback(tt-monkeys);

I understand it's confusing at first, and your wish to take everything one step at a time. But whatever you expect to validate or verify, it's not such of a big deal. Just try if calling an extension works, that's validation enough. Or why not just call from one phone to the other over the LAN? A second test is to see if your NAT works: call a SIP address from outside the LAN. (Call some SIP test numbers, for instance.)

[gatorback]

Thanks Okki, I used the SIP configuration above with the dialplan: