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.
[101]
type=friend
host=dynamic
nat=yes
qualify=yes
context=mario-default
defaultuser=101
secret=XXXXXXXXX
callerid="user1" <101>
mailbox=101
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?
(*xx.|*xxx|75xx|[3469]11|0|00|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.|**275x.)
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.
TAG: [GV-OPT]