[HELP] external call to sip phone not working


#1

Hello,

I have a DiD that is configured and has worked with an IAX2 client. I am now using X-Lite as a SIP phone and modified the extensions.conf file and sip.conf file accordingly (see below). When I try to call the SIP phone however, the following error is displayed in the CLI:

– Executing Dial(“IAX2/sixTel-1”, “SIP/home”) in new stack
Destroying call '3986193c29b75a3146f56155558ff439@127.0.0.1’
May 22 11:42:07 NOTICE[18503]: 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)

My extensions.conf file is:

exten => 5555556045,1,Answer()
exten => 5555556045,2,Wait(2)
exten => 5555556045,3,Dial(SIP/home)

My sip.conf file contains:
[home]
type=friend
secret=xxxxx
qualify=yes ; Qualify peer is no more than 2000 ms away
nat=no ; This phone is not natted
host=dynamic ; This device registers with us
canreinvite=yes ; Asterisk by default tries to redirect
context=internal ; the internal context controls what we can do
callerID="Kevin Williams <5555556045>"
mailbox=555@mb_tarity

This phone is on the same network as the asterisk server so NAT should not be an issue. Any idea what would cause this error message? Any pointers would be GREATLY appreciated.