Asterisk 1.8 is running on an ASUS RT-N16 (Optware) and I’m 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.
Q1) Do the two tests below validate the ATA to * connection?
Test 1) SPA2102 => Voice => Info shows that the device is registered.
Test 2) On the Asterisk side:
Name/username Host Dyn Forcerport ACL Port Status
101/101 192.168.8.110 D N 5060 [ur]OK (6 ms[/u])
NokiaE71x (Unspecified) D 0 Unmonitored
2 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 1 offline]
Do the underlined elements provide sufficient verifcation that the ATA and extension are properly configured to communicate with one another?
Q2) Will my ATA’s Dialplan prevent me from dialing out?
This dialplan was for a callcentric VOIP account.
Attempts to dial 18005551212 resulted a voice response ‘Enum lookup failed’. Attempts to dial 1(XXX)XXX-XXXX resulted in a busy signal. I realize that these observations are not shortcomings in the ATA - Asterisk configuration and are more likely to be a problem with Asterisk dialplan (extensions.conf) or GV (jabber.conf). For now, I would like to focus on validating the ATA-Asterisk connection.
If you have experience connecting an ATA to *, I would be interested in any constructive comments \ validation criteria. Thank you.