SIP phine works but can't dial into it


#1

I have a SIP phone. Using X-Lite. The phone calls out just fine but I can’t call into the phone. Asterisk is outside the NAT. The Laptop/Xlite is on the inside of a NAT.

extensions.conf

exten => 4,1,Dial(SIP/mperkel)

sip.conf

[mperkel]
type=friend
secret=*****
nat=route
host=dynamic
canreinvite=no
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
regexten=3333
callerid=“Marc Perkel” <9876543210>

When I cal in and try to ring the SIP phone I get this:

– Executing Dial(“IAX2/exgn-2”, “SIP/mperkel”) in new stack
Jan 25 14:20:22 NOTICE[567]: app_dial.c:1011 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel ‘IAX2/exgn-2’ status is ‘CHANUNAVAIL’

Running Asterisk 1.2.3

Thanks in advance