I’ve got a Cisco 7960 running the 7.4 SIP firmware along with a newly built Asterisk 1.4 server. I am able to place and recieve calls on x-lite softphones, including basic inbound and outbound dialing, but I seem to be missing something with my Cisco 7960.
Here is what the extension looks like in the extensions.conf:
Here is my SIPxxxxxx.conf for the phone (the config is being picked up and used by the phone):
# Line 1 Settings line1_name: "200" ; Line 1 Extension\User ID line1_displayname: "x200" ; Line 1 Display Name line1_shortname: "x200" ; Comment next to the button line1_authname: "200" ; Line 1 Registration Authentication line1_password: "8282" ; Line 1 Registration Password
And finally here is the extension in the sip.conf:
;x200 is the Cisco 7960 in the garage [x200] type=friend username=200 secret=8282 host=dynamic dtmfmode=rfc2833 qualify=500 nat=no canreinvite=no
The Cisco 7960 can send calls to any extension (including over my IAX2 trunk) just fine, it just won’t ring. The console output when I get the problem is:
[Jan 6 10:58:27] WARNING: app_dial.c:1081 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 'SIP/x202-081cbf60' status is 'CHANUNAVAIL'
What am I missing? Thanks in advance!