This is what I get from the asterisk CLI:
ast*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
202 (Unspecified) D 0 Unmonitored
201 (Unspecified) D 0 Unmonitored
2 sip peers [2 online , 0 offline]
ast*CLI> sip show users
Username Secret Accountcode
Def.Context ACL NAT
202 ******* from-sip
No RFC3581
201 ******* from-sip
No RFC3581
ast*CLI>
dave cantera wrote:
and
sip show users
Noah Miller wrote:
Hi Nathan -
No loop now, but instead I get this:
Mar 26 15:42:18 NOTICE[1854] app_dial.c: Unable to create channel of
type 'SIP' (cause 3 - No route to destination)
Mar 26 15:42:18 VERBOSE[1854] logger.c: == Everyone is busy/congested
at this time (1:0/0/1)
Mar 26 15:42:18 DEBUG[1854] app_dial.c: Exiting with
DIALSTATUS=CHANUNAVAIL.
Is the SIP device at 201 registered? What happens when you do a "sip
show peers"?
- Noah
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