[asterisk-users] Trouble with incoming calls

[James Caffrey]

Hello everyone. I am having trouble receiving via my Linksys SPA-3102. I
have not problem dialing out. It is like asterisk never even sees the call.
I have 3 sip devices grandstream bt-100, spa-3102 fxs, and spa-3102 fxo. A
very simple setup, just getting familar with asterisk. Here are my relative
config files. let me know if you need more.

sip.conf
[general]
context=default
bind=0.0.0.0
bindport=5060
srvlookup=yes

[100] ;bt-100
type=friend
username=100
context=default
secret=secret
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
mailbox=100 at default

[101] ;fxs
type=friend
username=pots
context=default
secret=phone
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
mailbox=101 at default

[102] ;fxo
type=friend
context=default
secret=pstn
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
port=5061

extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
context=default

[globals]
RINGGROUP1 => SIP/100&SIP/101

[default]
; These next three lines are for testing, just to make sure I got the call,
but no good
exten => s,1,Answer
exten => s,2,System(touch $HOME/got_it)
exten => s,3,Hangup
;exten => s,1,Dial(SIP/100,10)
;exten => s,2,Hangup
exten => 97,1,Dial(${RINGGROUP1},10)
exten => 97,n,Hangup
exten => 98,1,Answer
exten => 98,n,AGI(agi-test.agi)
exten => 98,n,Hangup
exten => 99,1,Answer
exten => 99,n,Playback(hello-world)
exten => 99,n,Hangup
exten => 100,1,Answer
exten => 100,n,Dial(SIP/100,15)
exten => 100,n,VoiceMail(100 at default)
exten => 100,n,Playback(vm-goodbye)
exten => 100,n,Hangup
exten => 101,1,Answer
exten => 101,n,Dial(SIP/101)
exten => 101,n,Hangup
exten => _XXXXXXXXXX,1,Dial(SIP/102/${EXTEN})
exten => _XXXXXXXXXX,n,Hangup

I appreciate your help

- Jim
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