[asterisk-users] Re: Polycom 601 loop
Noah Miller
noahisaacmiller at gmail.com
Mon Mar 26 14:57:23 MST 2007
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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